mirror of
https://github.com/FFmpeg/FFmpeg.git
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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
635 lines
24 KiB
C
635 lines
24 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/float_dsp.h"
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#include "libavutil/internal.h"
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#include "libavutil/mem.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 "filters.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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#define B0 0
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#define B1 1
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#define B2 2
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#define A1 3
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#define A2 4
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typedef struct BiquadCoeffs {
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double cd[5];
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float cf[5];
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} BiquadCoeffs;
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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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char *gains_str;
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int order_opt;
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float level_in;
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int precision;
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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[MAX_SPLITS];
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float gains[MAX_BANDS];
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BiquadCoeffs lp[MAX_BANDS][20];
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BiquadCoeffs hp[MAX_BANDS][20];
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BiquadCoeffs ap[MAX_BANDS][20];
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AVFrame *xover;
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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 filter order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, .unit = "m" },
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{ "2nd", "2nd order (12 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "m" },
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{ "4th", "4th order (24 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "m" },
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{ "6th", "6th order (36 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "m" },
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{ "8th", "8th order (48 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "m" },
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{ "10th", "10th order (60 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, .unit = "m" },
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{ "12th", "12th order (72 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, .unit = "m" },
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{ "14th", "14th order (84 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, .unit = "m" },
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{ "16th", "16th order (96 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, .unit = "m" },
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{ "18th", "18th order (108 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, .unit = "m" },
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{ "20th", "20th order (120 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, .unit = "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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{ "gain", "set output bands gain", OFFSET(gains_str), AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },
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{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
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{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
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{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
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{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(acrossover);
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static int query_formats(AVFilterContext *ctx)
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{
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AudioCrossoverContext *s = ctx->priv;
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static const enum AVSampleFormat auto_sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE
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};
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const enum AVSampleFormat *sample_fmts_list = sample_fmts;
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int ret = ff_set_common_all_channel_counts(ctx);
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if (ret < 0)
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return ret;
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switch (s->precision) {
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case 0:
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sample_fmts_list = auto_sample_fmts;
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break;
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case 1:
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sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
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break;
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case 2:
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sample_fmts[0] = AV_SAMPLE_FMT_DBLP;
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break;
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default:
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break;
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}
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ret = ff_set_common_formats_from_list(ctx, sample_fmts_list);
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if (ret < 0)
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return ret;
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return ff_set_common_all_samplerates(ctx);
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}
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static int parse_gains(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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saveptr = NULL;
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p = s->gains_str;
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for (i = 0; i < MAX_BANDS; i++) {
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float gain;
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char c[3] = { 0 };
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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%2s", &gain, c) < 1) {
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av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);
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ret = AVERROR(EINVAL);
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break;
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}
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if (c[0] == 'd' && c[1] == 'B')
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s->gains[i] = expf(gain * M_LN10 / 20.f);
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else
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s->gains[i] = gain;
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}
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for (; i < MAX_BANDS; i++)
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s->gains[i] = 1.f;
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return ret;
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}
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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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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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ret = parse_gains(ctx);
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if (ret < 0)
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return ret;
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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_append_outpad_free_name(ctx, &pad)) < 0)
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return ret;
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}
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return ret;
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}
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static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
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{
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double omega = 2. * 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->cd[B0] = b0 / a0;
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b->cd[B1] = b1 / a0;
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b->cd[B2] = b2 / a0;
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b->cd[A1] = -a1 / a0;
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b->cd[A2] = -a2 / a0;
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b->cf[B0] = b->cd[B0];
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b->cf[B1] = b->cd[B1];
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b->cf[B2] = b->cd[B2];
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b->cf[A1] = b->cd[A1];
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b->cf[A2] = b->cd[A2];
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}
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static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
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{
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double omega = 2. * 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->cd[B0] = b0 / a0;
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b->cd[B1] = b1 / a0;
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b->cd[B2] = b2 / a0;
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b->cd[A1] = -a1 / a0;
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b->cd[A2] = -a2 / a0;
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b->cf[B0] = b->cd[B0];
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b->cf[B1] = b->cd[B1];
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b->cf[B2] = b->cd[B2];
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b->cf[A1] = b->cd[A1];
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b->cf[A2] = b->cd[A2];
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}
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static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
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{
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double omega = 2. * 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->cd[B0] = b0 / a0;
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b->cd[B1] = b1 / a0;
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b->cd[B2] = b2 / a0;
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b->cd[A1] = -a1 / a0;
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b->cd[A2] = -a2 / a0;
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b->cf[B0] = b->cd[B0];
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b->cf[B1] = b->cd[B1];
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b->cf[B2] = b->cd[B2];
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b->cf[A1] = b->cd[A1];
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b->cf[A2] = b->cd[A2];
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}
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static void set_ap1(BiquadCoeffs *b, double fc, double sr)
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{
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double omega = 2. * M_PI * fc / sr;
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b->cd[A1] = exp(-omega);
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b->cd[A2] = 0.;
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b->cd[B0] = -b->cd[A1];
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b->cd[B1] = 1.;
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b->cd[B2] = 0.;
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b->cf[B0] = b->cd[B0];
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b->cf[B1] = b->cd[B1];
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b->cf[B2] = b->cd[B2];
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b->cf[A1] = b->cd[A1];
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b->cf[A2] = b->cd[A2];
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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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#define BIQUAD_PROCESS(name, type) \
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static void biquad_process_## name(const type *const c, \
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type *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 = c[B0]; \
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const type b1 = c[B1]; \
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const type b2 = c[B2]; \
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const type a1 = c[A1]; \
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const type a2 = c[A2]; \
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type z1 = b[0]; \
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type z2 = b[1]; \
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\
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for (int n = 0; n + 1 < nb_samples; n++) { \
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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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n++; \
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in = src[n]; \
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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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if (nb_samples & 1) { \
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const int n = nb_samples - 1; \
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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[0] = z1; \
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b[1] = 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 = arg; \
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AVFrame **frames = s->frames; \
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const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs; \
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const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; \
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const int nb_samples = in->nb_samples; \
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const int nb_outs = ctx->nb_outputs; \
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const int first_order = s->first_order; \
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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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type *xover = (type *)s->xover->extended_data[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, FFALIGN(nb_samples, sizeof(type))); \
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\
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for (int band = 0; band < nb_outs; band++) { \
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for (int f = 0; band + 1 < nb_outs && 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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type *hp = xover + nb_outs * 20 + band * 20 + f * 2; \
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const type *const hpc = (type *)&s->hp[band][f].c ## ff; \
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\
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biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \
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} \
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\
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for (int f = 0; band + 1 < nb_outs && 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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type *lp = xover + band * 20 + f * 2; \
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const type *const lpc = (type *)&s->lp[band][f].c ## ff; \
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\
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biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \
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} \
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\
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for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \
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if (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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type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \
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const type *const apc = (type *)&s->ap[aband][0].c ## ff; \
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\
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biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
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} \
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\
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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,
|
|
};
|