mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
8 years ago
parent
80bb81a8f3
commit
5d7c76566c
6 changed files with 710 additions and 2 deletions
@ -0,0 +1,689 @@ |
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/*
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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); |
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if (ret < 0) { |
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uninit(ctx); |
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return ret; |
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} |
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} |
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tstr2 = av_strtok(p2, " ", &saveptr2); |
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if (tstr2) { |
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sscanf(tstr2, "%lf", &s->bands[i].delay); |
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max_delay_size = FFMAX(max_delay_size, s->bands[i].delay * outlink->sample_rate); |
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tstr2 = av_strtok(p2, " ", &saveptr2); |
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if (tstr2) { |
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double initial_volume; |
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sscanf(tstr2, "%lf", &initial_volume); |
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initial_volume = pow(10.0, initial_volume / 20); |
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|
||||
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, |
||||
}; |
||||
Loading…
Reference in new issue