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avfilter: add chorus filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol
parent
415f1fab8d
commit
d2fc702ace
@ -16,6 +16,7 @@ version <next>:
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- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
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- WebM Live Chunk Muxer
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- nvenc level and tier options
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- chorus filter
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version 2.6:
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@ -1320,6 +1320,61 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
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side_right.wav
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@end example
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@section chorus
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Add a chorus effect to the audio.
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Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
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Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
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constant, with chorus, it is varied using using sinusoidal or triangular modulation.
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The modulation depth defines the range the modulated delay is played before or after
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the delay. Hence the delayed sound will sound slower or faster, that is the delayed
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sound tuned around the original one, like in a chorus where some vocals are slightly
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off key.
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It accepts the following parameters:
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@table @option
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@item in_gain
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Set input gain. Default is 0.4.
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@item out_gain
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Set output gain. Default is 0.4.
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@item delays
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Set delays. A typical delay is around 40ms to 60ms.
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@item decays
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Set decays.
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@item speeds
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Set speeds.
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@item depths
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Set depths.
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@end table
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@subsection Examples
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@itemize
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@item
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A single delay:
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@example
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chorus=0.7:0.9:55:0.4:0.25:2
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@end example
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@item
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Two delays:
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@example
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chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
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@end example
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@item
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Fuller sounding chorus with three delays:
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@example
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chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3
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@end example
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@end itemize
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@section compand
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Compress or expand the audio's dynamic range.
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@ -64,6 +64,7 @@ OBJS-$(CONFIG_BIQUAD_FILTER) += af_biquads.o
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OBJS-$(CONFIG_BS2B_FILTER) += af_bs2b.o
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OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
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OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
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OBJS-$(CONFIG_CHORUS_FILTER) += af_chorus.o generate_wave_table.o
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OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
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OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o
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OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
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379
libavfilter/af_chorus.c
Normal file
379
libavfilter/af_chorus.c
Normal file
@ -0,0 +1,379 @@
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/*
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* Copyright (c) 1998 Juergen Mueller And Sundry Contributors
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* This source code is freely redistributable and may be used for
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* any purpose. This copyright notice must be maintained.
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* Juergen Mueller And Sundry Contributors are not responsible for
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* the consequences of using this software.
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*
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* Copyright (c) 2015 Paul B Mahol
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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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* chorus audio filter
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*/
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#include "libavutil/avstring.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 "internal.h"
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#include "generate_wave_table.h"
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typedef struct ChorusContext {
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const AVClass *class;
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float in_gain, out_gain;
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char *delays_str;
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char *decays_str;
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char *speeds_str;
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char *depths_str;
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float *delays;
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float *decays;
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float *speeds;
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float *depths;
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uint8_t **chorusbuf;
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int **phase;
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int *length;
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int32_t **lookup_table;
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int *counter;
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int num_chorus;
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int max_samples;
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int channels;
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int modulation;
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int fade_out;
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int64_t next_pts;
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} ChorusContext;
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#define OFFSET(x) offsetof(ChorusContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption chorus_options[] = {
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{ "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
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{ "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
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{ "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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{ "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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{ "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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{ "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(chorus);
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static void count_items(char *item_str, int *nb_items)
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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 == '|')
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(*nb_items)++;
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}
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}
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static void fill_items(char *item_str, int *nb_items, float *items)
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{
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char *p, *saveptr = NULL;
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int i, new_nb_items = 0;
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p = item_str;
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for (i = 0; i < *nb_items; i++) {
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char *tstr = av_strtok(p, "|", &saveptr);
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p = NULL;
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new_nb_items += sscanf(tstr, "%f", &items[i]) == 1;
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}
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*nb_items = new_nb_items;
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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ChorusContext *s = ctx->priv;
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int nb_delays, nb_decays, nb_speeds, nb_depths;
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if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
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av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
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return AVERROR(EINVAL);
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}
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count_items(s->delays_str, &nb_delays);
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count_items(s->decays_str, &nb_decays);
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count_items(s->speeds_str, &nb_speeds);
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count_items(s->depths_str, &nb_depths);
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s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
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s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
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s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
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s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
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if (!s->delays || !s->decays || !s->speeds || !s->depths)
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return AVERROR(ENOMEM);
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fill_items(s->delays_str, &nb_delays, s->delays);
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fill_items(s->decays_str, &nb_decays, s->decays);
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fill_items(s->speeds_str, &nb_speeds, s->speeds);
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fill_items(s->depths_str, &nb_depths, s->depths);
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if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
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av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
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return AVERROR(EINVAL);
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}
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s->num_chorus = nb_delays;
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if (s->num_chorus < 1) {
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av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
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return AVERROR(EINVAL);
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}
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s->length = av_calloc(s->num_chorus, sizeof(*s->length));
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s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
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if (!s->length || !s->lookup_table)
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return AVERROR(ENOMEM);
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s->next_pts = AV_NOPTS_VALUE;
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_layouts();
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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 int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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ChorusContext *s = ctx->priv;
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float sum_in_volume = 1.0;
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int n;
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s->channels = outlink->channels;
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for (n = 0; n < s->num_chorus; n++) {
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int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
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int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
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s->length[n] = outlink->sample_rate / s->speeds[n];
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s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
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if (!s->lookup_table[n])
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return AVERROR(ENOMEM);
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ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
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s->length[n], 0., depth_samples, 0);
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s->max_samples = FFMAX(s->max_samples, samples);
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}
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for (n = 0; n < s->num_chorus; n++)
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sum_in_volume += s->decays[n];
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if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
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av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
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s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
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if (!s->counter)
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return AVERROR(ENOMEM);
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s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
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if (!s->phase)
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return AVERROR(ENOMEM);
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for (n = 0; n < outlink->channels; n++) {
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s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
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if (!s->phase[n])
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return AVERROR(ENOMEM);
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}
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s->fade_out = s->max_samples;
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return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
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outlink->channels,
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s->max_samples,
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outlink->format, 0);
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}
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#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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AVFilterContext *ctx = inlink->dst;
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ChorusContext *s = ctx->priv;
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AVFrame *out_frame;
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int c, i, n;
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if (av_frame_is_writable(frame)) {
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out_frame = frame;
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} else {
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out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
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if (!out_frame)
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return AVERROR(ENOMEM);
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av_frame_copy_props(out_frame, frame);
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}
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for (c = 0; c < inlink->channels; c++) {
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const float *src = (const float *)frame->extended_data[c];
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float *dst = (float *)out_frame->extended_data[c];
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float *chorusbuf = (float *)s->chorusbuf[c];
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int *phase = s->phase[c];
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for (i = 0; i < frame->nb_samples; i++) {
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float out, in = src[i];
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out = in * s->in_gain;
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for (n = 0; n < s->num_chorus; n++) {
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out += chorusbuf[MOD(s->max_samples + s->counter[c] -
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s->lookup_table[n][phase[n]],
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s->max_samples)] * s->decays[n];
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phase[n] = MOD(phase[n] + 1, s->length[n]);
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}
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out *= s->out_gain;
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dst[i] = out;
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chorusbuf[s->counter[c]] = in;
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s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
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}
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}
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s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
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if (frame != out_frame)
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av_frame_free(&frame);
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return ff_filter_frame(ctx->outputs[0], out_frame);
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}
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static int request_frame(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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ChorusContext *s = ctx->priv;
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int ret;
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ret = ff_request_frame(ctx->inputs[0]);
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if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
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int nb_samples = FFMIN(s->fade_out, 2048);
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AVFrame *frame;
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frame = ff_get_audio_buffer(outlink, nb_samples);
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if (!frame)
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return AVERROR(ENOMEM);
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s->fade_out -= nb_samples;
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av_samples_set_silence(frame->extended_data, 0,
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frame->nb_samples,
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outlink->channels,
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frame->format);
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frame->pts = s->next_pts;
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if (s->next_pts != AV_NOPTS_VALUE)
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s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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ret = filter_frame(ctx->inputs[0], frame);
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}
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return ret;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ChorusContext *s = ctx->priv;
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int n;
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av_freep(&s->delays);
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av_freep(&s->decays);
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av_freep(&s->speeds);
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av_freep(&s->depths);
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if (s->chorusbuf)
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av_freep(&s->chorusbuf[0]);
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av_freep(&s->chorusbuf);
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if (s->phase)
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for (n = 0; n < s->channels; n++)
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av_freep(&s->phase[n]);
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av_freep(&s->phase);
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av_freep(&s->counter);
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av_freep(&s->length);
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if (s->lookup_table)
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for (n = 0; n < s->num_chorus; n++)
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av_freep(&s->lookup_table[n]);
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av_freep(&s->lookup_table);
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}
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static const AVFilterPad chorus_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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{ NULL }
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};
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static const AVFilterPad chorus_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.request_frame = request_frame,
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.config_props = config_output,
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},
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{ NULL }
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};
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AVFilter ff_af_chorus = {
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.name = "chorus",
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.description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
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.query_formats = query_formats,
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.priv_size = sizeof(ChorusContext),
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.priv_class = &chorus_class,
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.init = init,
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.uninit = uninit,
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.inputs = chorus_inputs,
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.outputs = chorus_outputs,
|
||||
};
|
||||
@ -80,6 +80,7 @@ void avfilter_register_all(void)
|
||||
REGISTER_FILTER(BS2B, bs2b, af);
|
||||
REGISTER_FILTER(CHANNELMAP, channelmap, af);
|
||||
REGISTER_FILTER(CHANNELSPLIT, channelsplit, af);
|
||||
REGISTER_FILTER(CHORUS, chorus, af);
|
||||
REGISTER_FILTER(COMPAND, compand, af);
|
||||
REGISTER_FILTER(DCSHIFT, dcshift, af);
|
||||
REGISTER_FILTER(EARWAX, earwax, af);
|
||||
|
||||
@ -30,8 +30,8 @@
|
||||
#include "libavutil/version.h"
|
||||
|
||||
#define LIBAVFILTER_VERSION_MAJOR 5
|
||||
#define LIBAVFILTER_VERSION_MINOR 13
|
||||
#define LIBAVFILTER_VERSION_MICRO 101
|
||||
#define LIBAVFILTER_VERSION_MINOR 14
|
||||
#define LIBAVFILTER_VERSION_MICRO 100
|
||||
|
||||
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
|
||||
LIBAVFILTER_VERSION_MINOR, \
|
||||
|
||||
Loading…
Reference in New Issue
Block a user