generic-library/ffmpeg
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

avfilter: add adelay filter

Browse Source 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2013-09-13 11:36:52 +00:00
parent 42b8f5fba1
commit 9d05de2258
6 changed files with 315 additions and 1 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
Changelog
View File

@ -23,6 +23,8 @@ version <next>
- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
more consistent with other muxers.
- adelay filter
version 2.0:

27
doc/filters.texi
View File

@ -347,6 +347,33 @@ aconvert=u8:auto
@end example
@end itemize
@section adelay
Delay one or more audio channels.
Samples in delayed channel are filled with silence.
The filter accepts the following option:
@table @option
@item delays
Set list of delays in milliseconds for each channel separated by '|'.
At least one delay greater than 0 should be provided.
Unused delays will be silently ignored. If number of given delays is
smaller than number of channels all remaining channels will not be delayed.
@end table
@subsection Examples
@itemize
@item
Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave
the second channel (and any other channels that may be present) unchanged.
@example
adelay=1500:0:500
@end example
@end itemize
@section aecho
Apply echoing to the input audio.

1
libavfilter/Makefile
View File

@ -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

283
libavfilter/af_adelay.c Normal file
View File

@ -0,0 +1,283 @@
/*
* 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 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->index >= d->delay ? 0 : d->index; \
} \
} \
}
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");
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 must be specified.\n");
return AVERROR(EINVAL);
}
switch (inlink->format) {
case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ; break;
case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p; break;
case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p; break;
case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp; break;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AudioDelayContext *s = ctx->priv;
AVFrame *out_frame;
int i;
if (ctx->is_disabled || !s->delays)
return ff_filter_frame(ctx->outputs[0], frame);
out_frame = ff_get_audio_buffer(inlink, frame->nb_samples);
if (!out_frame)
return AVERROR(ENOMEM);
av_frame_copy_props(out_frame, frame);
for (i = 0; i < s->nb_delays; i++) {
ChanDelay *d = &s->chandelay[i];
const uint8_t *src = frame->extended_data[i];
uint8_t *dst = out_frame->extended_data[i];
if (!d->delay)
memcpy(dst, src, frame->nb_samples * s->block_align);
else
s->delay_channel(d, frame->nb_samples, src, dst);
}
s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
av_frame_free(&frame);
return ff_filter_frame(ctx->outputs[0], out_frame);
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioDelayContext *s = ctx->priv;
int ret;
ret = ff_request_frame(ctx->inputs[0]);
if (ret == AVERROR_EOF && !ctx->is_disabled && s->max_delay) {
int nb_samples = FFMIN(s->max_delay, 2048);
AVFrame *frame;
frame = ff_get_audio_buffer(outlink, nb_samples);
if (!frame)
return AVERROR(ENOMEM);
s->max_delay -= 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)
{
AudioDelayContext *s = ctx->priv;
int i;
for (i = 0; i < s->nb_delays; i++)
av_free(s->chandelay[i].samples);
av_freep(&s->chandelay);
}
static const AVFilterPad adelay_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad adelay_outputs[] = {
{
.name = "default",
.request_frame = request_frame,
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter avfilter_af_adelay = {
.name = "adelay",
.description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
.query_formats = query_formats,
.priv_size = sizeof(AudioDelayContext),
.priv_class = &adelay_class,
.uninit = uninit,
.inputs = adelay_inputs,
.outputs = adelay_outputs,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};

1
libavfilter/allfilters.c
View File

@ -48,6 +48,7 @@ void avfilter_register_all(void)
#if FF_API_ACONVERT_FILTER
REGISTER_FILTER(ACONVERT, aconvert, af);
#endif
REGISTER_FILTER(ADELAY, adelay, af);
REGISTER_FILTER(AECHO, aecho, af);
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFORMAT, aformat, af);

2
libavfilter/version.h
View File

@ -30,7 +30,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
#define LIBAVFILTER_VERSION_MINOR 84
#define LIBAVFILTER_VERSION_MINOR 85
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \