mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
1e7d2007c3
Makes it robust against adding fields before it, which will be useful in
following commits.
Majority of the patch generated by the following Coccinelle script:
@@
typedef AVOption;
identifier arr_name;
initializer list il;
initializer list[8] il1;
expression tail;
@@
AVOption arr_name[] = { il, { il1,
- tail
+ .unit = tail
}, ... };
with some manual changes, as the script:
* has trouble with options defined inside macros
* sometimes does not handle options under an #else branch
* sometimes swallows whitespace
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/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 "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; \
|
|
\
|
|
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,
|
|
};
|