499 lines
19 KiB
C
499 lines
19 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Crossover filter
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*
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* Split an audio stream into several bands.
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*/
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#include "libavutil/attributes.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/internal.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "internal.h"
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#define MAX_SPLITS 16
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#define MAX_BANDS MAX_SPLITS + 1
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typedef struct BiquadContext {
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double b0, b1, b2;
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double a1, a2;
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double z1, z2;
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} BiquadContext;
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typedef struct CrossoverChannel {
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BiquadContext lp[MAX_BANDS][20];
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BiquadContext hp[MAX_BANDS][20];
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BiquadContext ap[MAX_BANDS][MAX_BANDS][20];
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} CrossoverChannel;
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typedef struct AudioCrossoverContext {
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const AVClass *class;
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char *splits_str;
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int order_opt;
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float level_in;
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int order;
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int filter_count;
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int first_order;
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int ap_filter_count;
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int nb_splits;
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float *splits;
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CrossoverChannel *xover;
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AVFrame *input_frame;
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AVFrame *frames[MAX_BANDS];
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int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
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AVFloatDSPContext *fdsp;
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} AudioCrossoverContext;
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#define OFFSET(x) offsetof(AudioCrossoverContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption acrossover_options[] = {
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{ "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
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{ "order", "set order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, "m" },
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{ "2nd", "2nd order", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
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{ "4th", "4th order", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
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{ "6th", "6th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
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{ "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "m" },
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{ "10th", "10th order", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "m" },
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{ "12th", "12th order", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, "m" },
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{ "14th", "14th order", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, "m" },
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{ "16th", "16th order", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" },
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{ "18th", "18th order", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" },
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{ "20th", "20th order", 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" },
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{ "level", "set input gain", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(acrossover);
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioCrossoverContext *s = ctx->priv;
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char *p, *arg, *saveptr = NULL;
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int i, ret = 0;
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s->fdsp = avpriv_float_dsp_alloc(0);
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if (!s->fdsp)
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return AVERROR(ENOMEM);
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s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
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if (!s->splits)
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return AVERROR(ENOMEM);
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p = s->splits_str;
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for (i = 0; i < MAX_SPLITS; i++) {
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float freq;
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if (!(arg = av_strtok(p, " |", &saveptr)))
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break;
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p = NULL;
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if (av_sscanf(arg, "%f", &freq) != 1) {
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av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
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return AVERROR(EINVAL);
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}
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if (freq <= 0) {
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av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
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return AVERROR(EINVAL);
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}
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if (i > 0 && freq <= s->splits[i-1]) {
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av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
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return AVERROR(EINVAL);
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}
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s->splits[i] = freq;
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}
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s->nb_splits = i;
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for (i = 0; i <= s->nb_splits; i++) {
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AVFilterPad pad = { 0 };
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char *name;
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pad.type = AVMEDIA_TYPE_AUDIO;
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name = av_asprintf("out%d", ctx->nb_outputs);
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if (!name)
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return AVERROR(ENOMEM);
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pad.name = name;
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if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
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av_freep(&pad.name);
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return ret;
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}
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}
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return ret;
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}
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static void set_lp(BiquadContext *b, double fc, double q, double sr)
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{
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double omega = M_PI * fc / sr;
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double cosine = cos(omega);
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double alpha = sin(omega) / (2. * q);
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double b0 = (1. - cosine) / 2.;
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double b1 = 1. - cosine;
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double b2 = (1. - cosine) / 2.;
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double a0 = 1. + alpha;
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double a1 = -2. * cosine;
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double a2 = 1. - alpha;
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b->b0 = b0 / a0;
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b->b1 = b1 / a0;
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b->b2 = b2 / a0;
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b->a1 = -a1 / a0;
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b->a2 = -a2 / a0;
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}
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static void set_hp(BiquadContext *b, double fc, double q, double sr)
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{
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double omega = M_PI * fc / sr;
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double cosine = cos(omega);
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double alpha = sin(omega) / (2. * q);
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double b0 = (1. + cosine) / 2.;
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double b1 = -1. - cosine;
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double b2 = (1. + cosine) / 2.;
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double a0 = 1. + alpha;
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double a1 = -2. * cosine;
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double a2 = 1. - alpha;
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b->b0 = b0 / a0;
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b->b1 = b1 / a0;
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b->b2 = b2 / a0;
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b->a1 = -a1 / a0;
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b->a2 = -a2 / a0;
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}
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static void set_ap(BiquadContext *b, double fc, double q, double sr)
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{
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double omega = M_PI * fc / sr;
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double cosine = cos(omega);
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double alpha = sin(omega) / (2. * q);
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double a0 = 1. + alpha;
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double a1 = -2. * cosine;
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double a2 = 1. - alpha;
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double b0 = a2;
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double b1 = a1;
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double b2 = a0;
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b->b0 = b0 / a0;
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b->b1 = b1 / a0;
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b->b2 = b2 / a0;
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b->a1 = -a1 / a0;
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b->a2 = -a2 / a0;
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}
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static void set_ap1(BiquadContext *b, double fc, double sr)
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{
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double omega = M_PI * fc / sr;
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b->a1 = exp(-omega);
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b->a2 = 0.;
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b->b0 = -b->a1;
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b->b1 = 1.;
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b->b2 = 0.;
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}
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static void calc_q_factors(int order, double *q)
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{
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double n = order / 2.;
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for (int i = 0; i < n / 2; i++)
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q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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#define BIQUAD_PROCESS(name, type) \
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static void biquad_process_## name(BiquadContext *b, \
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type *dst, const type *src, \
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int nb_samples) \
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{ \
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const type b0 = b->b0; \
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const type b1 = b->b1; \
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const type b2 = b->b2; \
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const type a1 = b->a1; \
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const type a2 = b->a2; \
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type z1 = b->z1; \
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type z2 = b->z2; \
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\
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for (int n = 0; n < nb_samples; n++) { \
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const type in = src[n]; \
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type out; \
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\
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out = in * b0 + z1; \
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z1 = b1 * in + z2 + a1 * out; \
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z2 = b2 * in + a2 * out; \
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dst[n] = out; \
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} \
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\
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b->z1 = z1; \
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b->z2 = z2; \
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}
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BIQUAD_PROCESS(fltp, float)
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BIQUAD_PROCESS(dblp, double)
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#define XOVER_PROCESS(name, type, one, ff) \
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static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \
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{ \
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AudioCrossoverContext *s = ctx->priv; \
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AVFrame *in = s->input_frame; \
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AVFrame **frames = s->frames; \
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const int start = (in->channels * jobnr) / nb_jobs; \
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const int end = (in->channels * (jobnr+1)) / nb_jobs; \
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const int nb_samples = in->nb_samples; \
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\
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for (int ch = start; ch < end; ch++) { \
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const type *src = (const type *)in->extended_data[ch]; \
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CrossoverChannel *xover = &s->xover[ch]; \
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\
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s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \
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s->level_in, nb_samples); \
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emms_c(); \
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\
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for (int band = 0; band < ctx->nb_outputs; band++) { \
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for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
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const type *prv = (const type *)frames[band]->extended_data[ch]; \
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type *dst = (type *)frames[band + 1]->extended_data[ch]; \
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const type *hsrc = f == 0 ? prv : dst; \
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BiquadContext *hp = &xover->hp[band][f]; \
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\
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biquad_process_## name(hp, dst, hsrc, nb_samples); \
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} \
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\
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for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
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type *dst = (type *)frames[band]->extended_data[ch]; \
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const type *lsrc = dst; \
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BiquadContext *lp = &xover->lp[band][f]; \
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\
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biquad_process_## name(lp, dst, lsrc, nb_samples); \
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} \
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\
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for (int aband = band + 1; aband + 1 < ctx->nb_outputs; aband++) { \
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if (s->first_order) { \
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const type *asrc = (const type *)frames[band]->extended_data[ch]; \
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type *dst = (type *)frames[band]->extended_data[ch]; \
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BiquadContext *ap = &xover->ap[band][aband][0]; \
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\
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biquad_process_## name(ap, dst, asrc, nb_samples); \
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} \
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\
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for (int f = s->first_order; f < s->ap_filter_count; f++) { \
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const type *asrc = (const type *)frames[band]->extended_data[ch]; \
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type *dst = (type *)frames[band]->extended_data[ch]; \
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BiquadContext *ap = &xover->ap[band][aband][f]; \
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\
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biquad_process_## name(ap, dst, asrc, nb_samples); \
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} \
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} \
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} \
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\
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for (int band = 0; band < ctx->nb_outputs && s->first_order; band++) { \
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if (band & 1) { \
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type *dst = (type *)frames[band]->extended_data[ch]; \
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\
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for (int n = 0; n < nb_samples; n++) \
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dst[n] *= -one; \
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} \
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} \
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} \
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\
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return 0; \
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}
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XOVER_PROCESS(fltp, float, 1.f, f)
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XOVER_PROCESS(dblp, double, 1.0, d)
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioCrossoverContext *s = ctx->priv;
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int sample_rate = inlink->sample_rate;
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double q[16];
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s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
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if (!s->xover)
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return AVERROR(ENOMEM);
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s->order = (s->order_opt + 1) * 2;
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s->filter_count = s->order / 2;
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s->first_order = s->filter_count & 1;
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s->ap_filter_count = s->filter_count / 2 + s->first_order;
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calc_q_factors(s->order, q);
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for (int ch = 0; ch < inlink->channels; ch++) {
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for (int band = 0; band <= s->nb_splits; band++) {
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if (s->first_order) {
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set_lp(&s->xover[ch].lp[band][0], s->splits[band], 0.5, sample_rate);
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set_hp(&s->xover[ch].hp[band][0], s->splits[band], 0.5, sample_rate);
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}
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for (int n = s->first_order; n < s->filter_count; n++) {
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const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;
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set_lp(&s->xover[ch].lp[band][n], s->splits[band], q[idx], sample_rate);
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set_hp(&s->xover[ch].hp[band][n], s->splits[band], q[idx], sample_rate);
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}
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for (int x = 0; x <= s->nb_splits && s->first_order; x++)
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set_ap1(&s->xover[ch].ap[x][band][0], s->splits[band], sample_rate);
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for (int n = s->first_order; n < s->ap_filter_count; n++) {
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const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1);
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for (int x = 0; x <= s->nb_splits; x++)
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set_ap(&s->xover[ch].ap[x][band][n], s->splits[band], q[idx], sample_rate);
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}
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}
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}
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;
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case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
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}
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioCrossoverContext *s = ctx->priv;
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AVFrame **frames = s->frames;
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int i, ret = 0;
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for (i = 0; i < ctx->nb_outputs; i++) {
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frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
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if (!frames[i]) {
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ret = AVERROR(ENOMEM);
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break;
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}
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frames[i]->pts = in->pts;
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}
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if (ret < 0)
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goto fail;
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s->input_frame = in;
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ctx->internal->execute(ctx, s->filter_channels, NULL, NULL, FFMIN(inlink->channels,
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ff_filter_get_nb_threads(ctx)));
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for (i = 0; i < ctx->nb_outputs; i++) {
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ret = ff_filter_frame(ctx->outputs[i], frames[i]);
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frames[i] = NULL;
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if (ret < 0)
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break;
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}
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fail:
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for (i = 0; i < ctx->nb_outputs; i++)
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av_frame_free(&frames[i]);
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av_frame_free(&in);
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s->input_frame = NULL;
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return ret;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioCrossoverContext *s = ctx->priv;
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int i;
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av_freep(&s->fdsp);
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av_freep(&s->splits);
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av_freep(&s->xover);
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for (i = 0; i < ctx->nb_outputs; i++)
|
|
av_freep(&ctx->output_pads[i].name);
|
|
}
|
|
|
|
static const AVFilterPad inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = filter_frame,
|
|
.config_props = config_input,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
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,
|
|
.uninit = uninit,
|
|
.query_formats = query_formats,
|
|
.inputs = inputs,
|
|
.outputs = NULL,
|
|
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
|
|
AVFILTER_FLAG_SLICE_THREADS,
|
|
};
|