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avfilter: add adelay filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol
parent
42b8f5fba1
commit
9d05de2258
@ -23,6 +23,8 @@ version <next>
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- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
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- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
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more consistent with other muxers.
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- adelay filter
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version 2.0:
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@ -347,6 +347,33 @@ aconvert=u8:auto
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@end example
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@end itemize
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@section adelay
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Delay one or more audio channels.
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Samples in delayed channel are filled with silence.
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The filter accepts the following option:
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@table @option
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@item delays
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Set list of delays in milliseconds for each channel separated by '|'.
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At least one delay greater than 0 should be provided.
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Unused delays will be silently ignored. If number of given delays is
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smaller than number of channels all remaining channels will not be delayed.
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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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Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave
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the second channel (and any other channels that may be present) unchanged.
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@example
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adelay=1500:0:500
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@end example
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@end itemize
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@section aecho
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Apply echoing to the input audio.
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@ -52,6 +52,7 @@ OBJS-$(CONFIG_AVFORMAT) += lavfutils.o
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OBJS-$(CONFIG_SWSCALE) += lswsutils.o
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OBJS-$(CONFIG_ACONVERT_FILTER) += af_aconvert.o
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OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
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OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
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OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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283
libavfilter/af_adelay.c
Normal file
283
libavfilter/af_adelay.c
Normal file
@ -0,0 +1,283 @@
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/*
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* Copyright (c) 2013 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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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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typedef struct ChanDelay {
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int delay;
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unsigned delay_index;
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unsigned index;
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uint8_t *samples;
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} ChanDelay;
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typedef struct AudioDelayContext {
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const AVClass *class;
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char *delays;
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ChanDelay *chandelay;
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int nb_delays;
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int block_align;
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unsigned max_delay;
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int64_t next_pts;
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void (*delay_channel)(ChanDelay *d, int nb_samples,
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const uint8_t *src, uint8_t *dst);
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} AudioDelayContext;
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#define OFFSET(x) offsetof(AudioDelayContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption adelay_options[] = {
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{ "delays", "set list of delays for each channel", OFFSET(delays), 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(adelay);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterChannelLayouts *layouts;
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AVFilterFormats *formats;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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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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ff_set_common_channel_layouts(ctx, layouts);
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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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ff_set_common_formats(ctx, formats);
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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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ff_set_common_samplerates(ctx, formats);
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return 0;
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}
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#define DELAY(name, type, fill) \
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static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
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const uint8_t *ssrc, uint8_t *ddst) \
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{ \
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const type *src = (type *)ssrc; \
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type *dst = (type *)ddst; \
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type *samples = (type *)d->samples; \
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\
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while (nb_samples) { \
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if (d->delay_index < d->delay) { \
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const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
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\
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memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
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memset(dst, fill, len * sizeof(type)); \
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d->delay_index += len; \
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src += len; \
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dst += len; \
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nb_samples -= len; \
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} else { \
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*dst = samples[d->index]; \
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samples[d->index] = *src; \
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nb_samples--; \
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d->index++; \
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src++, dst++; \
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d->index = d->index >= d->delay ? 0 : d->index; \
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} \
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} \
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}
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DELAY(u8, uint8_t, 0x80)
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DELAY(s16, int16_t, 0)
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DELAY(s32, int32_t, 0)
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DELAY(flt, float, 0)
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DELAY(dbl, double, 0)
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioDelayContext *s = ctx->priv;
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char *p, *arg, *saveptr = NULL;
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int i;
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s->chandelay = av_calloc(inlink->channels, sizeof(*s->chandelay));
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if (!s->chandelay)
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return AVERROR(ENOMEM);
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s->nb_delays = inlink->channels;
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s->block_align = av_get_bytes_per_sample(inlink->format);
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p = s->delays;
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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float delay;
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if (!(arg = av_strtok(p, "|", &saveptr)))
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break;
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p = NULL;
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sscanf(arg, "%f", &delay);
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d->delay = delay * inlink->sample_rate / 1000.0;
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if (d->delay < 0) {
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av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
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return AVERROR(EINVAL);
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}
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}
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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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if (!d->delay)
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continue;
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d->samples = av_malloc_array(d->delay, s->block_align);
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if (!d->samples)
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return AVERROR(ENOMEM);
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s->max_delay = FFMAX(s->max_delay, d->delay);
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}
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if (!s->max_delay) {
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av_log(ctx, AV_LOG_ERROR, "At least one delay >0 must be specified.\n");
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return AVERROR(EINVAL);
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}
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switch (inlink->format) {
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case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
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case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
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case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
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case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
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case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
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}
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return 0;
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}
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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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AudioDelayContext *s = ctx->priv;
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AVFrame *out_frame;
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int i;
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if (ctx->is_disabled || !s->delays)
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return ff_filter_frame(ctx->outputs[0], frame);
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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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for (i = 0; i < s->nb_delays; i++) {
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ChanDelay *d = &s->chandelay[i];
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const uint8_t *src = frame->extended_data[i];
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uint8_t *dst = out_frame->extended_data[i];
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if (!d->delay)
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memcpy(dst, src, frame->nb_samples * s->block_align);
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else
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s->delay_channel(d, frame->nb_samples, src, dst);
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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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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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AudioDelayContext *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->max_delay) {
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int nb_samples = FFMIN(s->max_delay, 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->max_delay -= 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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AudioDelayContext *s = ctx->priv;
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int i;
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for (i = 0; i < s->nb_delays; i++)
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av_free(s->chandelay[i].samples);
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av_freep(&s->chandelay);
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}
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static const AVFilterPad adelay_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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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 adelay_outputs[] = {
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{
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.name = "default",
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.request_frame = request_frame,
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter avfilter_af_adelay = {
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.name = "adelay",
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.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
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.query_formats = query_formats,
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.priv_size = sizeof(AudioDelayContext),
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.priv_class = &adelay_class,
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.uninit = uninit,
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.inputs = adelay_inputs,
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.outputs = adelay_outputs,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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};
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@ -48,6 +48,7 @@ void avfilter_register_all(void)
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#if FF_API_ACONVERT_FILTER
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REGISTER_FILTER(ACONVERT, aconvert, af);
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#endif
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REGISTER_FILTER(ADELAY, adelay, af);
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REGISTER_FILTER(AECHO, aecho, af);
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REGISTER_FILTER(AFADE, afade, af);
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REGISTER_FILTER(AFORMAT, aformat, af);
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@ -30,7 +30,7 @@
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#include "libavutil/avutil.h"
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#define LIBAVFILTER_VERSION_MAJOR 3
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#define LIBAVFILTER_VERSION_MINOR 84
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#define LIBAVFILTER_VERSION_MINOR 85
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#define LIBAVFILTER_VERSION_MICRO 100
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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