mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
690 lines
21 KiB
C
690 lines
21 KiB
C
/*
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* COpyright (c) 2002 Daniel Pouzzner
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* Copyright (c) 1999 Chris Bagwell
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* Copyright (c) 1999 Nick Bailey
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* Copyright (c) 2007 Rob Sykes <robs@users.sourceforge.net>
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* Copyright (c) 2013 Paul B Mahol
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* Copyright (c) 2014 Andrew Kelley
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*
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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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* audio multiband compand filter
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/avstring.h"
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#include "libavutil/ffmath.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct CompandSegment {
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double x, y;
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double a, b;
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} CompandSegment;
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typedef struct CompandT {
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CompandSegment *segments;
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int nb_segments;
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double in_min_lin;
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double out_min_lin;
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double curve_dB;
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double gain_dB;
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} CompandT;
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#define N 4
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typedef struct PrevCrossover {
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double in;
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double out_low;
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double out_high;
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} PrevCrossover[N * 2];
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typedef struct Crossover {
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PrevCrossover *previous;
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size_t pos;
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double coefs[3 *(N+1)];
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} Crossover;
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typedef struct CompBand {
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CompandT transfer_fn;
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double *attack_rate;
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double *decay_rate;
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double *volume;
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double delay;
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double topfreq;
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Crossover filter;
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AVFrame *delay_buf;
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size_t delay_size;
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ptrdiff_t delay_buf_ptr;
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size_t delay_buf_cnt;
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} CompBand;
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typedef struct MCompandContext {
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const AVClass *class;
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char *args;
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int nb_bands;
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CompBand *bands;
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AVFrame *band_buf1, *band_buf2, *band_buf3;
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int band_samples;
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size_t delay_buf_size;
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} MCompandContext;
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#define OFFSET(x) offsetof(MCompandContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption mcompand_options[] = {
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{ "args", "set parameters for each band", OFFSET(args), AV_OPT_TYPE_STRING, { .str = "0.005,0.1 6 -47/-40,-34/-34,-17/-33 100 | 0.003,0.05 6 -47/-40,-34/-34,-17/-33 400 | 0.000625,0.0125 6 -47/-40,-34/-34,-15/-33 1600 | 0.0001,0.025 6 -47/-40,-34/-34,-31/-31,-0/-30 6400 | 0,0.025 6 -38/-31,-28/-28,-0/-25 22000" }, 0, 0, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(mcompand);
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static av_cold void uninit(AVFilterContext *ctx)
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{
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MCompandContext *s = ctx->priv;
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int i;
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av_frame_free(&s->band_buf1);
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av_frame_free(&s->band_buf2);
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av_frame_free(&s->band_buf3);
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if (s->bands) {
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for (i = 0; i < s->nb_bands; i++) {
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av_freep(&s->bands[i].attack_rate);
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av_freep(&s->bands[i].decay_rate);
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av_freep(&s->bands[i].volume);
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av_freep(&s->bands[i].transfer_fn.segments);
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av_freep(&s->bands[i].filter.previous);
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av_frame_free(&s->bands[i].delay_buf);
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}
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}
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av_freep(&s->bands);
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterChannelLayouts *layouts;
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AVFilterFormats *formats;
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static const enum AVSampleFormat sample_fmts[] = {
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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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static void count_items(char *item_str, int *nb_items, char delimiter)
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{
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char *p;
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*nb_items = 1;
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for (p = item_str; *p; p++) {
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if (*p == delimiter)
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(*nb_items)++;
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}
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}
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static void update_volume(CompBand *cb, double in, int ch)
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{
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double delta = in - cb->volume[ch];
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if (delta > 0.0)
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cb->volume[ch] += delta * cb->attack_rate[ch];
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else
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cb->volume[ch] += delta * cb->decay_rate[ch];
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}
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static double get_volume(CompandT *s, double in_lin)
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{
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CompandSegment *cs;
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double in_log, out_log;
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int i;
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if (in_lin <= s->in_min_lin)
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return s->out_min_lin;
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in_log = log(in_lin);
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for (i = 1; i < s->nb_segments; i++)
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if (in_log <= s->segments[i].x)
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break;
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cs = &s->segments[i - 1];
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in_log -= cs->x;
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out_log = cs->y + in_log * (cs->a * in_log + cs->b);
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return exp(out_log);
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}
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static int parse_points(char *points, int nb_points, double radius,
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CompandT *s, AVFilterContext *ctx)
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{
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int new_nb_items, num;
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char *saveptr = NULL;
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char *p = points;
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int i;
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#define S(x) s->segments[2 * ((x) + 1)]
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for (i = 0, new_nb_items = 0; i < nb_points; i++) {
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char *tstr = av_strtok(p, ",", &saveptr);
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p = NULL;
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if (!tstr || sscanf(tstr, "%lf/%lf", &S(i).x, &S(i).y) != 2) {
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av_log(ctx, AV_LOG_ERROR,
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"Invalid and/or missing input/output value.\n");
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return AVERROR(EINVAL);
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}
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if (i && S(i - 1).x > S(i).x) {
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av_log(ctx, AV_LOG_ERROR,
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"Transfer function input values must be increasing.\n");
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return AVERROR(EINVAL);
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}
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S(i).y -= S(i).x;
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av_log(ctx, AV_LOG_DEBUG, "%d: x=%f y=%f\n", i, S(i).x, S(i).y);
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new_nb_items++;
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}
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num = new_nb_items;
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/* Add 0,0 if necessary */
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if (num == 0 || S(num - 1).x)
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num++;
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#undef S
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#define S(x) s->segments[2 * (x)]
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/* Add a tail off segment at the start */
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S(0).x = S(1).x - 2 * s->curve_dB;
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S(0).y = S(1).y;
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num++;
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/* Join adjacent colinear segments */
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for (i = 2; i < num; i++) {
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double g1 = (S(i - 1).y - S(i - 2).y) * (S(i - 0).x - S(i - 1).x);
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double g2 = (S(i - 0).y - S(i - 1).y) * (S(i - 1).x - S(i - 2).x);
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int j;
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if (fabs(g1 - g2))
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continue;
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num--;
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for (j = --i; j < num; j++)
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S(j) = S(j + 1);
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}
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for (i = 0; i < s->nb_segments; i += 2) {
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s->segments[i].y += s->gain_dB;
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s->segments[i].x *= M_LN10 / 20;
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s->segments[i].y *= M_LN10 / 20;
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}
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#define L(x) s->segments[i - (x)]
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for (i = 4; i < s->nb_segments; i += 2) {
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double x, y, cx, cy, in1, in2, out1, out2, theta, len, r;
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L(4).a = 0;
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L(4).b = (L(2).y - L(4).y) / (L(2).x - L(4).x);
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L(2).a = 0;
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L(2).b = (L(0).y - L(2).y) / (L(0).x - L(2).x);
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theta = atan2(L(2).y - L(4).y, L(2).x - L(4).x);
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len = hypot(L(2).x - L(4).x, L(2).y - L(4).y);
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r = FFMIN(radius, len);
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L(3).x = L(2).x - r * cos(theta);
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L(3).y = L(2).y - r * sin(theta);
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theta = atan2(L(0).y - L(2).y, L(0).x - L(2).x);
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len = hypot(L(0).x - L(2).x, L(0).y - L(2).y);
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r = FFMIN(radius, len / 2);
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x = L(2).x + r * cos(theta);
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y = L(2).y + r * sin(theta);
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cx = (L(3).x + L(2).x + x) / 3;
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cy = (L(3).y + L(2).y + y) / 3;
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L(2).x = x;
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L(2).y = y;
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in1 = cx - L(3).x;
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out1 = cy - L(3).y;
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in2 = L(2).x - L(3).x;
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out2 = L(2).y - L(3).y;
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L(3).a = (out2 / in2 - out1 / in1) / (in2 - in1);
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L(3).b = out1 / in1 - L(3).a * in1;
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}
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L(3).x = 0;
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L(3).y = L(2).y;
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s->in_min_lin = exp(s->segments[1].x);
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s->out_min_lin = exp(s->segments[1].y);
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return 0;
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}
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static void square_quadratic(double const *x, double *y)
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{
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y[0] = x[0] * x[0];
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y[1] = 2 * x[0] * x[1];
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y[2] = 2 * x[0] * x[2] + x[1] * x[1];
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y[3] = 2 * x[1] * x[2];
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y[4] = x[2] * x[2];
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}
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static int crossover_setup(AVFilterLink *outlink, Crossover *p, double frequency)
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{
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double w0 = 2 * M_PI * frequency / outlink->sample_rate;
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double Q = sqrt(.5), alpha = sin(w0) / (2*Q);
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double x[9], norm;
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int i;
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if (w0 > M_PI)
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return AVERROR(EINVAL);
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x[0] = (1 - cos(w0))/2; /* Cf. filter_LPF in biquads.c */
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x[1] = 1 - cos(w0);
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x[2] = (1 - cos(w0))/2;
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x[3] = (1 + cos(w0))/2; /* Cf. filter_HPF in biquads.c */
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x[4] = -(1 + cos(w0));
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x[5] = (1 + cos(w0))/2;
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x[6] = 1 + alpha;
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x[7] = -2*cos(w0);
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x[8] = 1 - alpha;
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for (norm = x[6], i = 0; i < 9; ++i)
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x[i] /= norm;
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square_quadratic(x , p->coefs);
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square_quadratic(x + 3, p->coefs + 5);
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square_quadratic(x + 6, p->coefs + 10);
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p->previous = av_calloc(outlink->channels, sizeof(*p->previous));
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if (!p->previous)
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return AVERROR(ENOMEM);
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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MCompandContext *s = ctx->priv;
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int ret, ch, i, k, new_nb_items, nb_bands;
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char *p = s->args, *saveptr = NULL;
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int max_delay_size = 0;
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count_items(s->args, &nb_bands, '|');
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s->nb_bands = FFMAX(1, nb_bands);
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s->bands = av_calloc(nb_bands, sizeof(*s->bands));
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if (!s->bands)
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return AVERROR(ENOMEM);
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for (i = 0, new_nb_items = 0; i < nb_bands; i++) {
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int nb_points, nb_attacks, nb_items = 0;
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char *tstr2, *tstr = av_strtok(p, "|", &saveptr);
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char *p2, *p3, *saveptr2 = NULL, *saveptr3 = NULL;
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double radius;
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if (!tstr) {
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uninit(ctx);
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return AVERROR(EINVAL);
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}
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p = NULL;
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p2 = tstr;
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count_items(tstr, &nb_items, ' ');
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tstr2 = av_strtok(p2, " ", &saveptr2);
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if (!tstr2) {
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av_log(ctx, AV_LOG_ERROR, "at least one attacks/decays rate is mandatory\n");
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uninit(ctx);
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return AVERROR(EINVAL);
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}
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p2 = NULL;
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p3 = tstr2;
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count_items(tstr2, &nb_attacks, ',');
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if (!nb_attacks || nb_attacks & 1) {
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av_log(ctx, AV_LOG_ERROR, "number of attacks rate plus decays rate must be even\n");
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uninit(ctx);
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return AVERROR(EINVAL);
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}
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s->bands[i].attack_rate = av_calloc(outlink->channels, sizeof(double));
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s->bands[i].decay_rate = av_calloc(outlink->channels, sizeof(double));
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s->bands[i].volume = av_calloc(outlink->channels, sizeof(double));
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for (k = 0; k < FFMIN(nb_attacks / 2, outlink->channels); k++) {
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char *tstr3 = av_strtok(p3, ",", &saveptr3);
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p3 = NULL;
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sscanf(tstr3, "%lf", &s->bands[i].attack_rate[k]);
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tstr3 = av_strtok(p3, ",", &saveptr3);
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sscanf(tstr3, "%lf", &s->bands[i].decay_rate[k]);
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if (s->bands[i].attack_rate[k] > 1.0 / outlink->sample_rate) {
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s->bands[i].attack_rate[k] = 1.0 - exp(-1.0 / (outlink->sample_rate * s->bands[i].attack_rate[k]));
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} else {
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s->bands[i].attack_rate[k] = 1.0;
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}
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if (s->bands[i].decay_rate[k] > 1.0 / outlink->sample_rate) {
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s->bands[i].decay_rate[k] = 1.0 - exp(-1.0 / (outlink->sample_rate * s->bands[i].decay_rate[k]));
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} else {
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s->bands[i].decay_rate[k] = 1.0;
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}
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}
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for (ch = k; ch < outlink->channels; ch++) {
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s->bands[i].attack_rate[ch] = s->bands[i].attack_rate[k - 1];
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s->bands[i].decay_rate[ch] = s->bands[i].decay_rate[k - 1];
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}
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tstr2 = av_strtok(p2, " ", &saveptr2);
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if (!tstr2) {
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av_log(ctx, AV_LOG_ERROR, "transfer function curve in dB must be set\n");
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uninit(ctx);
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return AVERROR(EINVAL);
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}
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sscanf(tstr2, "%lf", &s->bands[i].transfer_fn.curve_dB);
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radius = s->bands[i].transfer_fn.curve_dB * M_LN10 / 20.0;
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tstr2 = av_strtok(p2, " ", &saveptr2);
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if (!tstr2) {
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av_log(ctx, AV_LOG_ERROR, "transfer points missing\n");
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uninit(ctx);
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return AVERROR(EINVAL);
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}
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count_items(tstr2, &nb_points, ',');
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s->bands[i].transfer_fn.nb_segments = (nb_points + 4) * 2;
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s->bands[i].transfer_fn.segments = av_calloc(s->bands[i].transfer_fn.nb_segments,
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sizeof(CompandSegment));
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if (!s->bands[i].transfer_fn.segments) {
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uninit(ctx);
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return AVERROR(ENOMEM);
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}
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ret = parse_points(tstr2, nb_points, radius, &s->bands[i].transfer_fn, ctx);
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if (ret < 0) {
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av_log(ctx, AV_LOG_ERROR, "transfer points parsing failed\n");
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uninit(ctx);
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return ret;
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}
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tstr2 = av_strtok(p2, " ", &saveptr2);
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if (!tstr2) {
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av_log(ctx, AV_LOG_ERROR, "crossover_frequency is missing\n");
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uninit(ctx);
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return AVERROR(EINVAL);
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}
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new_nb_items += sscanf(tstr2, "%lf", &s->bands[i].topfreq) == 1;
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if (s->bands[i].topfreq < 0 || s->bands[i].topfreq >= outlink->sample_rate / 2) {
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av_log(ctx, AV_LOG_ERROR, "crossover_frequency should be >=0 and lower than half of sample rate\n");
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uninit(ctx);
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return AVERROR(EINVAL);
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}
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if (s->bands[i].topfreq != 0) {
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ret = crossover_setup(outlink, &s->bands[i].filter, s->bands[i].topfreq);
|
|
if (ret < 0) {
|
|
uninit(ctx);
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
tstr2 = av_strtok(p2, " ", &saveptr2);
|
|
if (tstr2) {
|
|
sscanf(tstr2, "%lf", &s->bands[i].delay);
|
|
max_delay_size = FFMAX(max_delay_size, s->bands[i].delay * outlink->sample_rate);
|
|
|
|
tstr2 = av_strtok(p2, " ", &saveptr2);
|
|
if (tstr2) {
|
|
double initial_volume;
|
|
|
|
sscanf(tstr2, "%lf", &initial_volume);
|
|
initial_volume = pow(10.0, initial_volume / 20);
|
|
|
|
for (k = 0; k < outlink->channels; k++) {
|
|
s->bands[i].volume[k] = initial_volume;
|
|
}
|
|
|
|
tstr2 = av_strtok(p2, " ", &saveptr2);
|
|
if (tstr2) {
|
|
sscanf(tstr2, "%lf", &s->bands[i].transfer_fn.gain_dB);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
s->nb_bands = new_nb_items;
|
|
|
|
for (i = 0; max_delay_size > 0 && i < s->nb_bands; i++) {
|
|
s->bands[i].delay_buf = ff_get_audio_buffer(outlink, max_delay_size);
|
|
if (!s->bands[i].delay_buf)
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
s->delay_buf_size = max_delay_size;
|
|
|
|
return 0;
|
|
}
|
|
|
|
#define CONVOLVE _ _ _ _
|
|
|
|
static void crossover(int ch, Crossover *p,
|
|
double *ibuf, double *obuf_low,
|
|
double *obuf_high, size_t len)
|
|
{
|
|
double out_low, out_high;
|
|
|
|
while (len--) {
|
|
p->pos = p->pos ? p->pos - 1 : N - 1;
|
|
#define _ out_low += p->coefs[j] * p->previous[ch][p->pos + j].in \
|
|
- p->coefs[2*N+2 + j] * p->previous[ch][p->pos + j].out_low, j++;
|
|
{
|
|
int j = 1;
|
|
out_low = p->coefs[0] * *ibuf;
|
|
CONVOLVE
|
|
*obuf_low++ = out_low;
|
|
}
|
|
#undef _
|
|
#define _ out_high += p->coefs[j+N+1] * p->previous[ch][p->pos + j].in \
|
|
- p->coefs[2*N+2 + j] * p->previous[ch][p->pos + j].out_high, j++;
|
|
{
|
|
int j = 1;
|
|
out_high = p->coefs[N+1] * *ibuf;
|
|
CONVOLVE
|
|
*obuf_high++ = out_high;
|
|
}
|
|
p->previous[ch][p->pos + N].in = p->previous[ch][p->pos].in = *ibuf++;
|
|
p->previous[ch][p->pos + N].out_low = p->previous[ch][p->pos].out_low = out_low;
|
|
p->previous[ch][p->pos + N].out_high = p->previous[ch][p->pos].out_high = out_high;
|
|
}
|
|
}
|
|
|
|
static int mcompand_channel(MCompandContext *c, CompBand *l, double *ibuf, double *obuf, int len, int ch)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < len; i++) {
|
|
double level_in_lin, level_out_lin, checkbuf;
|
|
/* Maintain the volume fields by simulating a leaky pump circuit */
|
|
update_volume(l, fabs(ibuf[i]), ch);
|
|
|
|
/* Volume memory is updated: perform compand */
|
|
level_in_lin = l->volume[ch];
|
|
level_out_lin = get_volume(&l->transfer_fn, level_in_lin);
|
|
|
|
if (c->delay_buf_size <= 0) {
|
|
checkbuf = ibuf[i] * level_out_lin;
|
|
obuf[i] = checkbuf;
|
|
} else {
|
|
double *delay_buf = (double *)l->delay_buf->extended_data[ch];
|
|
|
|
/* FIXME: note that this lookahead algorithm is really lame:
|
|
the response to a peak is released before the peak
|
|
arrives. */
|
|
|
|
/* because volume application delays differ band to band, but
|
|
total delay doesn't, the volume is applied in an iteration
|
|
preceding that in which the sample goes to obuf, except in
|
|
the band(s) with the longest vol app delay.
|
|
|
|
the offset between delay_buf_ptr and the sample to apply
|
|
vol to, is a constant equal to the difference between this
|
|
band's delay and the longest delay of all the bands. */
|
|
|
|
if (l->delay_buf_cnt >= l->delay_size) {
|
|
checkbuf =
|
|
delay_buf[(l->delay_buf_ptr +
|
|
c->delay_buf_size -
|
|
l->delay_size) % c->delay_buf_size] * level_out_lin;
|
|
delay_buf[(l->delay_buf_ptr + c->delay_buf_size -
|
|
l->delay_size) % c->delay_buf_size] = checkbuf;
|
|
}
|
|
if (l->delay_buf_cnt >= c->delay_buf_size) {
|
|
obuf[i] = delay_buf[l->delay_buf_ptr];
|
|
} else {
|
|
l->delay_buf_cnt++;
|
|
}
|
|
delay_buf[l->delay_buf_ptr++] = ibuf[i];
|
|
l->delay_buf_ptr %= c->delay_buf_size;
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
MCompandContext *s = ctx->priv;
|
|
AVFrame *out, *abuf, *bbuf, *cbuf;
|
|
int ch, band, i;
|
|
|
|
out = ff_get_audio_buffer(outlink, in->nb_samples);
|
|
if (!out) {
|
|
av_frame_free(&in);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
if (s->band_samples < in->nb_samples) {
|
|
av_frame_free(&s->band_buf1);
|
|
av_frame_free(&s->band_buf2);
|
|
av_frame_free(&s->band_buf3);
|
|
|
|
s->band_buf1 = ff_get_audio_buffer(outlink, in->nb_samples);
|
|
s->band_buf2 = ff_get_audio_buffer(outlink, in->nb_samples);
|
|
s->band_buf3 = ff_get_audio_buffer(outlink, in->nb_samples);
|
|
s->band_samples = in->nb_samples;
|
|
}
|
|
|
|
for (ch = 0; ch < outlink->channels; ch++) {
|
|
double *a, *dst = (double *)out->extended_data[ch];
|
|
|
|
for (band = 0, abuf = in, bbuf = s->band_buf2, cbuf = s->band_buf1; band < s->nb_bands; band++) {
|
|
CompBand *b = &s->bands[band];
|
|
|
|
if (b->topfreq) {
|
|
crossover(ch, &b->filter, (double *)abuf->extended_data[ch],
|
|
(double *)bbuf->extended_data[ch], (double *)cbuf->extended_data[ch], in->nb_samples);
|
|
} else {
|
|
bbuf = abuf;
|
|
abuf = cbuf;
|
|
}
|
|
|
|
if (abuf == in)
|
|
abuf = s->band_buf3;
|
|
mcompand_channel(s, b, (double *)bbuf->extended_data[ch], (double *)abuf->extended_data[ch], out->nb_samples, ch);
|
|
a = (double *)abuf->extended_data[ch];
|
|
for (i = 0; i < out->nb_samples; i++) {
|
|
dst[i] += a[i];
|
|
}
|
|
|
|
FFSWAP(AVFrame *, abuf, cbuf);
|
|
}
|
|
}
|
|
|
|
out->pts = in->pts;
|
|
av_frame_free(&in);
|
|
return ff_filter_frame(outlink, out);
|
|
}
|
|
|
|
static int request_frame(AVFilterLink *outlink)
|
|
{
|
|
AVFilterContext *ctx = outlink->src;
|
|
int ret;
|
|
|
|
ret = ff_request_frame(ctx->inputs[0]);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static const AVFilterPad mcompand_inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.filter_frame = filter_frame,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad mcompand_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.request_frame = request_frame,
|
|
.config_props = config_output,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
|
|
AVFilter ff_af_mcompand = {
|
|
.name = "mcompand",
|
|
.description = NULL_IF_CONFIG_SMALL(
|
|
"Multiband Compress or expand audio dynamic range."),
|
|
.query_formats = query_formats,
|
|
.priv_size = sizeof(MCompandContext),
|
|
.priv_class = &mcompand_class,
|
|
.uninit = uninit,
|
|
.inputs = mcompand_inputs,
|
|
.outputs = mcompand_outputs,
|
|
};
|