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lavfi: add aecho filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2013-07-08 13:44:35 +00:00
parent 4977e467a5
commit 884c890535
6 changed files with 423 additions and 2 deletions
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2
Changelog
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@ -3,6 +3,8 @@ releases are sorted from youngest to oldest.
version <next>
- aecho filter
version 2.0:

60
doc/filters.texi
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@ -347,6 +347,66 @@ aconvert=u8:auto
@end example
@end itemize
@section aecho
Apply echoing to the input audio.
Echoes are reflected sound and can occur naturally amongst mountains
(and sometimes large buildings) when talking or shouting; digital echo
effects emulate this behaviour and are often used to help fill out the
sound of a single instrument or vocal. The time difference between the
original signal and the reflection is the @code{delay}, and the
loudness of the reflected signal is the @code{decay}.
Multiple echoes can have different delays and decays.
A description of the accepted parameters follows.
@table @option
@item in_gain
Set input gain of reflected signal. Default is @code{0.6}.
@item out_gain
Set output gain of reflected signal. Default is @code{0.3}.
@item delays
Set list of time intervals in milliseconds between original signal and reflections
separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}.
Default is @code{1000}.
@item decays
Set list of loudnesses of reflected signals separated by '|'.
Allowed range for each @code{decay} is @code{(0 - 1.0]}.
Default is @code{0.5}.
@end table
@subsection Examples
@itemize
@item
Make it sound as if there are twice as many instruments as are actually playing:
@example
aecho=0.8:0.88:60:0.4
@end example
@item
If delay is very short, then it sound like a (metallic) robot playing music:
@example
aecho=0.8:0.88:6:0.4
@end example
@item
A longer delay will sound like an open air concert in the mountains:
@example
aecho=0.8:0.9:1000:0.3
@end example
@item
Same as above but with one more mountain:
@example
aecho=0.8:0.9:1000|1800:0.3|0.25
@end example
@end itemize
@section afade
Apply fade-in/out effect to input audio.

1
libavfilter/Makefile
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@ -52,6 +52,7 @@ OBJS-$(CONFIG_AVFORMAT) += lavfutils.o
OBJS-$(CONFIG_SWSCALE) += lswsutils.o
OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o

357
libavfilter/af_aecho.c Normal file
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@ -0,0 +1,357 @@
/*
* Copyright (c) 2013 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
*
*/
#include "libavutil/avstring.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"
typedef struct AudioEchoContext {
const AVClass *class;
float in_gain, out_gain;
char *delays, *decays;
float *delay, *decay;
int nb_echoes;
int delay_index;
uint8_t **delayptrs;
int max_samples, fade_out;
int *samples;
int64_t next_pts;
void (*echo_samples)(struct AudioEchoContext *ctx, uint8_t **delayptrs,
uint8_t * const *src, uint8_t **dst,
int nb_samples, int channels);
} AudioEchoContext;
#define OFFSET(x) offsetof(AudioEchoContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption aecho_options[] = {
{ "in_gain", "set signal input gain", OFFSET(in_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.6}, 0, 1, A },
{ "out_gain", "set signal output gain", OFFSET(out_gain), AV_OPT_TYPE_FLOAT, {.dbl=0.3}, 0, 1, A },
{ "delays", "set list of signal delays", OFFSET(delays), AV_OPT_TYPE_STRING, {.str="1000"}, 0, 0, A },
{ "decays", "set list of signal decays", OFFSET(decays), AV_OPT_TYPE_STRING, {.str="0.5"}, 0, 0, A },
{ NULL },
};
AVFILTER_DEFINE_CLASS(aecho);
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 void uninit(AVFilterContext *ctx)
{
AudioEchoContext *s = ctx->priv;
av_freep(&s->delay);
av_freep(&s->decay);
av_freep(&s->samples);
if (s->delayptrs)
av_freep(s->delayptrs[0]);
av_freep(&s->delayptrs);
}
static av_cold int init(AVFilterContext *ctx)
{
AudioEchoContext *s = ctx->priv;
int nb_delays, nb_decays, i;
if (!s->delays || !s->decays) {
av_log(ctx, AV_LOG_ERROR, "Missing delays and/or decays.\n");
return AVERROR(EINVAL);
}
count_items(s->delays, &nb_delays);
count_items(s->decays, &nb_decays);
s->delay = av_realloc_f(s->delay, nb_delays, sizeof(*s->delay));
s->decay = av_realloc_f(s->decay, nb_decays, sizeof(*s->decay));
if (!s->delay || !s->decay)
return AVERROR(ENOMEM);
fill_items(s->delays, &nb_delays, s->delay);
fill_items(s->decays, &nb_decays, s->decay);
if (nb_delays != nb_decays) {
av_log(ctx, AV_LOG_ERROR, "Number of delays %d differs from number of decays %d.\n", nb_delays, nb_decays);
return AVERROR(EINVAL);
}
s->nb_echoes = nb_delays;
if (!s->nb_echoes) {
av_log(ctx, AV_LOG_ERROR, "At least one decay & delay must be set.\n");
return AVERROR(EINVAL);
}
s->samples = av_realloc_f(s->samples, nb_delays, sizeof(*s->samples));
if (!s->samples)
return AVERROR(ENOMEM);
for (i = 0; i < nb_delays; i++) {
if (s->delay[i] <= 0 || s->delay[i] > 90000) {
av_log(ctx, AV_LOG_ERROR, "delay[%d]: %f is out of allowed range: (0, 90000]\n", i, s->delay[i]);
return AVERROR(EINVAL);
}
if (s->decay[i] <= 0 || s->decay[i] > 1) {
av_log(ctx, AV_LOG_ERROR, "decay[%d]: %f is out of allowed range: (0, 1]\n", i, s->decay[i]);
return AVERROR(EINVAL);
}
}
s->next_pts = AV_NOPTS_VALUE;
av_log(ctx, AV_LOG_DEBUG, "nb_echoes:%d\n", s->nb_echoes);
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layouts;
AVFilterFormats *formats;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ff_set_common_channel_layouts(ctx, layouts);
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_formats(ctx, formats);
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_samplerates(ctx, formats);
return 0;
}
#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
#define ECHO(name, type, min, max) \
static void echo_samples_## name ##p(AudioEchoContext *ctx, \
uint8_t **delayptrs, \
uint8_t * const *src, uint8_t **dst, \
int nb_samples, int channels) \
{ \
const double out_gain = ctx->out_gain; \
const double in_gain = ctx->in_gain; \
const int nb_echoes = ctx->nb_echoes; \
const int max_samples = ctx->max_samples; \
int i, j, chan, index; \
\
for (chan = 0; chan < channels; chan++) { \
const type *s = (type *)src[chan]; \
type *d = (type *)dst[chan]; \
type *dbuf = (type *)delayptrs[chan]; \
\
index = ctx->delay_index; \
for (i = 0; i < nb_samples; i++, s++, d++) { \
double out, in; \
\
in = *s; \
out = in * in_gain; \
for (j = 0; j < nb_echoes; j++) { \
int ix = index + max_samples - ctx->samples[j]; \
ix = MOD(ix, max_samples); \
out += dbuf[ix] * ctx->decay[j]; \
} \
out *= out_gain; \
\
*d = av_clipd(out, min, max); \
dbuf[index] = in; \
\
index = MOD(index + 1, max_samples); \
} \
} \
ctx->delay_index = index; \
}
ECHO(dbl, double, -1.0, 1.0 )
ECHO(flt, float, -1.0, 1.0 )
ECHO(s16, int16_t, INT16_MIN, INT16_MAX)
ECHO(s32, int32_t, INT32_MIN, INT32_MAX)
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioEchoContext *s = ctx->priv;
float volume = 1.0;
int i;
for (i = 0; i < s->nb_echoes; i++) {
s->samples[i] = s->delay[i] * outlink->sample_rate / 1000.0;
s->max_samples = FFMAX(s->max_samples, s->samples[i]);
volume += s->decay[i];
}
if (s->max_samples <= 0) {
av_log(ctx, AV_LOG_ERROR, "Nothing to echo - missing delay samples.\n");
return AVERROR(EINVAL);
}
s->fade_out = s->max_samples;
if (volume * s->in_gain * s->out_gain > 1.0)
av_log(ctx, AV_LOG_WARNING,
"out_gain %f can cause saturation of output\n", s->out_gain);
switch (outlink->format) {
case AV_SAMPLE_FMT_DBLP: s->echo_samples = echo_samples_dblp; break;
case AV_SAMPLE_FMT_FLTP: s->echo_samples = echo_samples_fltp; break;
case AV_SAMPLE_FMT_S16P: s->echo_samples = echo_samples_s16p; break;
case AV_SAMPLE_FMT_S32P: s->echo_samples = echo_samples_s32p; break;
}
if (s->delayptrs)
av_freep(s->delayptrs[0]);
av_freep(&s->delayptrs);
return av_samples_alloc_array_and_samples(&s->delayptrs, NULL,
outlink->channels,
s->max_samples,
outlink->format, 0);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AudioEchoContext *s = ctx->priv;
AVFrame *out_frame;
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);
}
s->echo_samples(s, s->delayptrs, frame->data, out_frame->data,
frame->nb_samples, inlink->channels);
if (frame != out_frame)
av_frame_free(&frame);
s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
return ff_filter_frame(ctx->outputs[0], out_frame);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioEchoContext *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);
s->echo_samples(s, s->delayptrs, frame->data, frame->data,
frame->nb_samples, outlink->channels);
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);
return ff_filter_frame(outlink, frame);
}
return ret;
}
static const AVFilterPad aecho_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL },
};
static const AVFilterPad aecho_outputs[] = {
{
.name = "default",
.request_frame = request_frame,
.config_props = config_output,
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL },
};
AVFilter avfilter_af_aecho = {
.name = "aecho",
.description = NULL_IF_CONFIG_SMALL("Add echoing to the audio."),
.query_formats = query_formats,
.priv_size = sizeof(AudioEchoContext),
.priv_class = &aecho_class,
.init = init,
.uninit = uninit,
.inputs = aecho_inputs,
.outputs = aecho_outputs,
};

1
libavfilter/allfilters.c
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@ -48,6 +48,7 @@ void avfilter_register_all(void)
#if FF_API_ACONVERT_FILTER
REGISTER_FILTER(ACONVERT, aconvert, af);
#endif
REGISTER_FILTER(AECHO, aecho, af);
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFORMAT, aformat, af);
REGISTER_FILTER(AINTERLEAVE, ainterleave, af);

4
libavfilter/version.h
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@ -30,8 +30,8 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
#define LIBAVFILTER_VERSION_MINOR 79
#define LIBAVFILTER_VERSION_MICRO 101
#define LIBAVFILTER_VERSION_MINOR 80
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \