[FFmpeg-devel] [PATCH] avfilter: add adelay filter
Stefano Sabatini
stefasab at gmail.com
Sun Sep 15 11:15:54 CEST 2013
On date Friday 2013-09-13 17:42:08 +0000, Paul B Mahol encoded:
> Signed-off-by: Paul B Mahol <onemda at gmail.com>
> ---
> doc/filters.texi | 15 +++
> libavfilter/Makefile | 1 +
> libavfilter/af_adelay.c | 296 +++++++++++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 313 insertions(+)
> create mode 100644 libavfilter/af_adelay.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 7f8d1b2..d4cec8a 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -347,6 +347,21 @@ aconvert=u8:auto
> @end example
> @end itemize
>
> + at section adelay
> +
> +Delay one or more audio channels.
> +
> +The filter accepts the following option:
> +
Please specify what happens when an audio channels is delayed (I
suppose it is filled with silence).
> + at table @option
> + at item delays
> +Set list of delays in milliseconds for each channel.
> +At least one delay greater than 0 should be provided.
> +Unused delays will be silently ignored. If number
> +of given delays is smaller than numer of channels all
> +remaining channels will be un-delayed.
Missing separator declaration. Also I wonder if it makes sense to
specify time durations instead.
> + at end table
> +
> @section aecho
>
> Apply echoing to the input audio.
> diff --git a/libavfilter/Makefile b/libavfilter/Makefile
> index b57d4c9..5a82c84 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_ADELAY_FILTER) += af_adelay.o
> OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
> OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
> OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
> diff --git a/libavfilter/af_adelay.c b/libavfilter/af_adelay.c
> new file mode 100644
> index 0000000..b74ddaf
> --- /dev/null
> +++ b/libavfilter/af_adelay.c
> @@ -0,0 +1,296 @@
> +/*
> + * 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 "avfilter.h"
> +#include "audio.h"
> +#include "internal.h"
> +
> +typedef struct ChanDelay {
> + int delay;
> + unsigned delay_index;
> + unsigned index;
> + uint8_t *samples;
> +} ChanDelay;
> +
> +typedef struct AudioDelayContext {
> + const AVClass *class;
> + char *delays;
> + ChanDelay *chandelay;
> + int nb_delays;
> + int block_align;
> + unsigned max_delay;
> + int64_t next_pts;
> +
> + void (*delay_channel)(ChanDelay *d, int nb_samples,
> + const uint8_t *src, uint8_t *dst);
> +} AudioDelayContext;
> +
> +#define OFFSET(x) offsetof(AudioDelayContext, x)
> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
> +
> +static const AVOption adelay_options[] = {
> + { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A },
> + { NULL }
> +};
> +
> +AVFILTER_DEFINE_CLASS(adelay);
> +
> +static av_cold int init(AVFilterContext *ctx)
> +{
> + AudioDelayContext *s = ctx->priv;
> +
> + if (!s->delays) {
> + av_log(ctx, AV_LOG_ERROR, "Missing delays.\n");
Nit: no need for final point (no complete sentence)
> + return AVERROR(EINVAL);
or maybe it could work a as a no-op (simplify scripting sometimes).
> + }
> +
> + return 0;
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> + AVFilterChannelLayouts *layouts;
> + AVFilterFormats *formats;
> + static const enum AVSampleFormat sample_fmts[] = {
> + AV_SAMPLE_FMT_U8P, 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 DELAY(name, type, fill) \
> +static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
> + const uint8_t *ssrc, uint8_t *ddst) \
> +{ \
> + const type *src = (type *)ssrc; \
> + type *dst = (type *)ddst; \
> + type *samples = (type *)d->samples; \
> + \
> + while (nb_samples) { \
> + if (d->delay_index < d->delay) { \
> + const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
> + \
> + memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
> + memset(dst, fill, len * sizeof(type)); \
> + d->delay_index += len; \
> + src += len; \
> + dst += len; \
> + nb_samples -= len; \
> + } else { \
> + *dst = samples[d->index]; \
> + samples[d->index] = *src; \
> + nb_samples--; \
> + d->index++; \
> + src++, dst++; \
> + d->index %= d->delay; \
> + } \
> + } \
> +}
> +
> +DELAY(u8, uint8_t, 0x80)
> +DELAY(s16, int16_t, 0)
> +DELAY(s32, int32_t, 0)
> +DELAY(flt, float, 0)
> +DELAY(dbl, double, 0)
> +
> +static int config_input(AVFilterLink *inlink)
> +{
> + AVFilterContext *ctx = inlink->dst;
> + AudioDelayContext *s = ctx->priv;
> + char *p, *arg, *saveptr = NULL;
> + int i;
> +
> + s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
> + if (!s->chandelay)
> + return AVERROR(ENOMEM);
> + s->nb_delays = inlink->channels;
> + s->block_align = av_get_bytes_per_sample(inlink->format);
> +
> + p = s->delays;
> + for (i = 0; i < s->nb_delays; i++) {
> + ChanDelay *d = &s->chandelay[i];
> + float delay;
> +
> + if (!(arg = av_strtok(p, "|", &saveptr)))
> + break;
> +
> + p = NULL;
> + sscanf(arg, "%f", &delay);
> +
> + d->delay = delay * inlink->sample_rate / 1000.0;
> + if (d->delay < 0) {
> + av_log(ctx, AV_LOG_ERROR, "Delay must be non-negative number.\n");
Nit: a non negative number
Same remark about milliseconds vs. time duration specification.
Also: would it make sense to specify a negative delay?
> + return AVERROR(EINVAL);
> + }
> + }
> +
> + for (i = 0; i < s->nb_delays; i++) {
> + ChanDelay *d = &s->chandelay[i];
> +
> + if (!d->delay)
> + continue;
> +
> + d->samples = av_malloc_array(d->delay, s->block_align);
> + if (!d->samples)
> + return AVERROR(ENOMEM);
> +
> + s->max_delay = FFMAX(s->max_delay, d->delay);
> + }
> +
> + if (!s->max_delay) {
> + av_log(ctx, AV_LOG_ERROR, "At least one delay >0 needed.\n");
Nit: is needed / must be specified.
[...]
--
FFmpeg = Fiendish and Fancy Murdering Power Exxagerate Goblin
More information about the ffmpeg-devel
mailing list