[FFmpeg-devel] [PATCH] lavfi: add aecho filter
Stefano Sabatini
stefasab at gmail.com
Wed Jul 10 10:59:12 CEST 2013
On date Tuesday 2013-07-09 23:13:59 +0000, Paul B Mahol encoded:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> doc/filters.texi | 60 ++++++++
> libavfilter/Makefile | 1 +
> libavfilter/af_aecho.c | 357 +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 419 insertions(+)
> create mode 100644 libavfilter/af_aecho.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 33436ad..92f8612 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -347,6 +347,66 @@ aconvert=u8:auto
> @end example
> @end itemize
>
> + at 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.
> +
> + at table @option
> + at item in_gain
> +Set input gain of reflected signal. Default is @code{0.6}.
> +
> + at item out_gain
> +Set output gain of reflected signal. Default is @code{0.3}.
This could be more explicit.
> +
> + at 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}.
> +
> + at 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}.
> + at end table
> +
> + at subsection Examples
> +
> + at itemize
> + at item
> +Make it sound as if there are twice as many instruments as are actually playing:
> + at example
> +aecho=0.8:0.88:60:0.4
> + at end example
> +
> + at item
> +If delay is very short, then it sound like a (metallic) robot playing music:
> + at example
> +aecho=0.8:0.88:6:0.4
> + at end example
> +
> + at item
> +A longer delay will sound like an open air concert in the mountains:
> + at example
> +aecho=0.8:0.9:1000:0.3
> + at end example
> +
> + at item
> +Same as above but with one more mountain:
> + at example
> +aecho=0.8:0.9:1000|1800:0.3|0.25
> + at end example
> + at end itemize
> +
> @section afade
>
> Apply fade-in/out effect to input audio.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index cf76ee1..306b24c 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -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
> diff --git a/libavfilter/af_aecho.c b/libavfilter/af_aecho.c
> new file mode 100644
> index 0000000..23158f7
> --- /dev/null
> +++ b/libavfilter/af_aecho.c
> @@ -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;
av_strtod()?
> + }
> +
> + *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");
Nit: Missing
First char is usually Capitalized, same below.
> + 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 differs from number of decays\n");
Number ... of delays %d differs from number of decays %d
Alternatively you can adopt a syntax which eliminates the problem, for
example:
12%34|56%78|...
> + 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);
> + }
Shouldn't work as a null filter in this case? You could just return a
warning (unless there are compelling reasons for not doing it).
[...]
> +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,
No timeline support?
--
FFmpeg = Fabulous and Fast Maxi Patchable Elitist Gem
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