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avfilter: add chorus filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2014-06-24 08:35:37 +00:00
parent 415f1fab8d
commit d2fc702ace
6 changed files with 439 additions and 2 deletions
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1
Changelog
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@ -16,6 +16,7 @@ version <next>:
- unpack DivX-style packed B-frames in MPEG-4 bitstream filter
- WebM Live Chunk Muxer
- nvenc level and tier options
- chorus filter
version 2.6:

55
doc/filters.texi
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@ -1320,6 +1320,61 @@ front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
side_right.wav
@end example
@section chorus
Add a chorus effect to the audio.
Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
constant, with chorus, it is varied using using sinusoidal or triangular modulation.
The modulation depth defines the range the modulated delay is played before or after
the delay. Hence the delayed sound will sound slower or faster, that is the delayed
sound tuned around the original one, like in a chorus where some vocals are slightly
off key.
It accepts the following parameters:
@table @option
@item in_gain
Set input gain. Default is 0.4.
@item out_gain
Set output gain. Default is 0.4.
@item delays
Set delays. A typical delay is around 40ms to 60ms.
@item decays
Set decays.
@item speeds
Set speeds.
@item depths
Set depths.
@end table
@subsection Examples
@itemize
@item
A single delay:
@example
chorus=0.7:0.9:55:0.4:0.25:2
@end example
@item
Two delays:
@example
chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
@end example
@item
Fuller sounding chorus with three delays:
@example
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
@end example
@end itemize
@section compand
Compress or expand the audio's dynamic range.

1
libavfilter/Makefile
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@ -64,6 +64,7 @@ OBJS-$(CONFIG_BIQUAD_FILTER) += af_biquads.o
OBJS-$(CONFIG_BS2B_FILTER) += af_bs2b.o
OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
OBJS-$(CONFIG_CHORUS_FILTER) += af_chorus.o generate_wave_table.o
OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o

379
libavfilter/af_chorus.c Normal file
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@ -0,0 +1,379 @@
/*
* Copyright (c) 1998 Juergen Mueller And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Juergen Mueller And Sundry Contributors are not responsible for
* the consequences of using this software.
*
* Copyright (c) 2015 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* chorus audio filter
*/
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
#include "generate_wave_table.h"
typedef struct ChorusContext {
const AVClass *class;
float in_gain, out_gain;
char *delays_str;
char *decays_str;
char *speeds_str;
char *depths_str;
float *delays;
float *decays;
float *speeds;
float *depths;
uint8_t **chorusbuf;
int **phase;
int *length;
int32_t **lookup_table;
int *counter;
int num_chorus;
int max_samples;
int channels;
int modulation;
int fade_out;
int64_t next_pts;
} ChorusContext;
#define OFFSET(x) offsetof(ChorusContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption chorus_options[] = {
{ "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
{ "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=.4}, 0, 1, A },
{ "delays", "set delays", OFFSET(delays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
{ "decays", "set decays", OFFSET(decays_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
{ "speeds", "set speeds", OFFSET(speeds_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
{ "depths", "set depths", OFFSET(depths_str), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(chorus);
static void count_items(char *item_str, int *nb_items)
{
char *p;
*nb_items = 1;
for (p = item_str; *p; p++) {
if (*p == '|')
(*nb_items)++;
}
}
static void fill_items(char *item_str, int *nb_items, float *items)
{
char *p, *saveptr = NULL;
int i, new_nb_items = 0;
p = item_str;
for (i = 0; i < *nb_items; i++) {
char *tstr = av_strtok(p, "|", &saveptr);
p = NULL;
new_nb_items += sscanf(tstr, "%f", &items[i]) == 1;
}
*nb_items = new_nb_items;
}
static av_cold int init(AVFilterContext *ctx)
{
ChorusContext *s = ctx->priv;
int nb_delays, nb_decays, nb_speeds, nb_depths;
if (!s->delays_str || !s->decays_str || !s->speeds_str || !s->depths_str) {
av_log(ctx, AV_LOG_ERROR, "Both delays & decays & speeds & depths must be set.\n");
return AVERROR(EINVAL);
}
count_items(s->delays_str, &nb_delays);
count_items(s->decays_str, &nb_decays);
count_items(s->speeds_str, &nb_speeds);
count_items(s->depths_str, &nb_depths);
s->delays = av_realloc_f(s->delays, nb_delays, sizeof(*s->delays));
s->decays = av_realloc_f(s->decays, nb_decays, sizeof(*s->decays));
s->speeds = av_realloc_f(s->speeds, nb_speeds, sizeof(*s->speeds));
s->depths = av_realloc_f(s->depths, nb_depths, sizeof(*s->depths));
if (!s->delays || !s->decays || !s->speeds || !s->depths)
return AVERROR(ENOMEM);
fill_items(s->delays_str, &nb_delays, s->delays);
fill_items(s->decays_str, &nb_decays, s->decays);
fill_items(s->speeds_str, &nb_speeds, s->speeds);
fill_items(s->depths_str, &nb_depths, s->depths);
if (nb_delays != nb_decays && nb_delays != nb_speeds && nb_delays != nb_depths) {
av_log(ctx, AV_LOG_ERROR, "Number of delays & decays & speeds & depths given must be same.\n");
return AVERROR(EINVAL);
}
s->num_chorus = nb_delays;
if (s->num_chorus < 1) {
av_log(ctx, AV_LOG_ERROR, "At least one delay & decay & speed & depth must be set.\n");
return AVERROR(EINVAL);
}
s->length = av_calloc(s->num_chorus, sizeof(*s->length));
s->lookup_table = av_calloc(s->num_chorus, sizeof(*s->lookup_table));
if (!s->length || !s->lookup_table)
return AVERROR(ENOMEM);
s->next_pts = AV_NOPTS_VALUE;
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
};
int ret;
layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ChorusContext *s = ctx->priv;
float sum_in_volume = 1.0;
int n;
s->channels = outlink->channels;
for (n = 0; n < s->num_chorus; n++) {
int samples = (int) ((s->delays[n] + s->depths[n]) * outlink->sample_rate / 1000.0);
int depth_samples = (int) (s->depths[n] * outlink->sample_rate / 1000.0);
s->length[n] = outlink->sample_rate / s->speeds[n];
s->lookup_table[n] = av_malloc(sizeof(int32_t) * s->length[n]);
if (!s->lookup_table[n])
return AVERROR(ENOMEM);
ff_generate_wave_table(WAVE_SIN, AV_SAMPLE_FMT_S32, s->lookup_table[n],
s->length[n], 0., depth_samples, 0);
s->max_samples = FFMAX(s->max_samples, samples);
}
for (n = 0; n < s->num_chorus; n++)
sum_in_volume += s->decays[n];
if (s->in_gain * (sum_in_volume) > 1.0 / s->out_gain)
av_log(ctx, AV_LOG_WARNING, "output gain can cause saturation or clipping of output\n");
s->counter = av_calloc(outlink->channels, sizeof(*s->counter));
if (!s->counter)
return AVERROR(ENOMEM);
s->phase = av_calloc(outlink->channels, sizeof(*s->phase));
if (!s->phase)
return AVERROR(ENOMEM);
for (n = 0; n < outlink->channels; n++) {
s->phase[n] = av_calloc(s->num_chorus, sizeof(int));
if (!s->phase[n])
return AVERROR(ENOMEM);
}
s->fade_out = s->max_samples;
return av_samples_alloc_array_and_samples(&s->chorusbuf, NULL,
outlink->channels,
s->max_samples,
outlink->format, 0);
}
#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
ChorusContext *s = ctx->priv;
AVFrame *out_frame;
int c, i, n;
if (av_frame_is_writable(frame)) {
out_frame = frame;
} else {
out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
if (!out_frame)
return AVERROR(ENOMEM);
av_frame_copy_props(out_frame, frame);
}
for (c = 0; c < inlink->channels; c++) {
const float *src = (const float *)frame->extended_data[c];
float *dst = (float *)out_frame->extended_data[c];
float *chorusbuf = (float *)s->chorusbuf[c];
int *phase = s->phase[c];
for (i = 0; i < frame->nb_samples; i++) {
float out, in = src[i];
out = in * s->in_gain;
for (n = 0; n < s->num_chorus; n++) {
out += chorusbuf[MOD(s->max_samples + s->counter[c] -
s->lookup_table[n][phase[n]],
s->max_samples)] * s->decays[n];
phase[n] = MOD(phase[n] + 1, s->length[n]);
}
out *= s->out_gain;
dst[i] = out;
chorusbuf[s->counter[c]] = in;
s->counter[c] = MOD(s->counter[c] + 1, s->max_samples);
}
}
s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
if (frame != out_frame)
av_frame_free(&frame);
return ff_filter_frame(ctx->outputs[0], out_frame);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ChorusContext *s = ctx->priv;
int ret;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF && !ctx->is_disabled && s->fade_out) {
int nb_samples = FFMIN(s->fade_out, 2048);
AVFrame *frame;
frame = ff_get_audio_buffer(outlink, nb_samples);
if (!frame)
return AVERROR(ENOMEM);
s->fade_out -= nb_samples;
av_samples_set_silence(frame->extended_data, 0,
frame->nb_samples,
outlink->channels,
frame->format);
frame->pts = s->next_pts;
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += av_rescale_q(nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
ret = filter_frame(ctx->inputs[0], frame);
}
return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
{
ChorusContext *s = ctx->priv;
int n;
av_freep(&s->delays);
av_freep(&s->decays);
av_freep(&s->speeds);
av_freep(&s->depths);
if (s->chorusbuf)
av_freep(&s->chorusbuf[0]);
av_freep(&s->chorusbuf);
if (s->phase)
for (n = 0; n < s->channels; n++)
av_freep(&s->phase[n]);
av_freep(&s->phase);
av_freep(&s->counter);
av_freep(&s->length);
if (s->lookup_table)
for (n = 0; n < s->num_chorus; n++)
av_freep(&s->lookup_table[n]);
av_freep(&s->lookup_table);
}
static const AVFilterPad chorus_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad chorus_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.request_frame = request_frame,
.config_props = config_output,
},
{ NULL }
};
AVFilter ff_af_chorus = {
.name = "chorus",
.description = NULL_IF_CONFIG_SMALL("Add a chorus effect to the audio."),
.query_formats = query_formats,
.priv_size = sizeof(ChorusContext),
.priv_class = &chorus_class,
.init = init,
.uninit = uninit,
.inputs = chorus_inputs,
.outputs = chorus_outputs,
};

1
libavfilter/allfilters.c
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@ -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);

4
libavfilter/version.h
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@ -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, \