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aphaser filter 

Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2013-03-30 04:03:39 +00:00
parent 83e2217221
commit bf65752848
6 changed files with 402 additions and 2 deletions
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1
Changelog
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@ -15,6 +15,7 @@ version <next>:
- new ffmpeg options -filter_script and -filter_complex_script, which allow a
filtergraph description to be read from a file
- OpenCL support
- audio phaser filter
version 1.2:

37
doc/filters.texi
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@ -6266,6 +6266,43 @@ following one, the permission might not be received as expected in that
following filter. Inserting a @ref{format} or @ref{aformat} filter before the
perms/aperms filter can avoid this problem.
@section aphaser
Add a phasing effect to the input audio.
A phaser filter creates series of peaks and troughs in the frequency spectrum.
The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.
The filter accepts parameters as a list of @var{key}=@var{value}
pairs, separated by ":".
A description of the accepted parameters follows.
@table @option
@item in_gain
Set input gain. Default is 0.4.
@item out_gain
Set output gain. Default is 0.74
@item delay
Set delay in milliseconds. Default is 3.0.
@item decay
Set decay. Default is 0.4.
@item speed
Set modulation speed in Hz. Default is 0.5.
@item type
Set modulation type. Default is triangular.
It accepts the following values:
@table @samp
@item triangular, t
@item sinusoidal, s
@end table
@end table
@section aselect, select
Select frames to pass in output.

1
libavfilter/Makefile
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@ -58,6 +58,7 @@ OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_ASELECT_FILTER) += f_select.o
OBJS-$(CONFIG_ASENDCMD_FILTER) += f_sendcmd.o

360
libavfilter/af_aphaser.c Normal file
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@ -0,0 +1,360 @@
/*
* 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
*/
/**
* @file
* phaser audio filter
*/
#include "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
enum WaveType {
WAVE_SIN,
WAVE_TRI,
WAVE_NB,
};
typedef struct AudioPhaserContext {
const AVClass *class;
double in_gain, out_gain;
double delay;
double decay;
double speed;
enum WaveType type;
int delay_buffer_length;
double *delay_buffer;
int modulation_buffer_length;
int32_t *modulation_buffer;
int delay_pos, modulation_pos;
void (*phaser)(struct AudioPhaserContext *p,
uint8_t * const *src, uint8_t **dst,
int nb_samples, int channels);
} AudioPhaserContext;
#define OFFSET(x) offsetof(AudioPhaserContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption aphaser_options[] = {
{ "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
{ "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
{ "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
{ "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
{ "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
{ "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
{ NULL },
};
AVFILTER_DEFINE_CLASS(aphaser);
static av_cold int init(AVFilterContext *ctx, const char *args)
{
AudioPhaserContext *p = ctx->priv;
if (p->in_gain > (1 - p->decay * p->decay))
av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
if (p->in_gain / (1 - p->decay) > 1 / p->out_gain)
av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
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;
}
static void generate_wave_table(enum WaveType wave_type, enum AVSampleFormat sample_fmt,
void *table, int table_size,
double min, double max, double phase)
{
uint32_t i, phase_offset = phase / M_PI / 2 * table_size + 0.5;
for (i = 0; i < table_size; i++) {
uint32_t point = (i + phase_offset) % table_size;
double d;
switch (wave_type) {
case WAVE_SIN:
d = (sin((double)point / table_size * 2 * M_PI) + 1) / 2;
break;
case WAVE_TRI:
d = (double)point * 2 / table_size;
switch (4 * point / table_size) {
case 0: d = d + 0.5; break;
case 1:
case 2: d = 1.5 - d; break;
case 3: d = d - 1.5; break;
}
break;
default:
av_assert0(0);
}
d = d * (max - min) + min;
switch (sample_fmt) {
case AV_SAMPLE_FMT_FLT: {
float *fp = (float *)table;
*fp++ = (float)d;
table = fp;
continue; }
case AV_SAMPLE_FMT_DBL: {
double *dp = (double *)table;
*dp++ = d;
table = dp;
continue; }
}
d += d < 0 ? -0.5 : 0.5;
switch (sample_fmt) {
case AV_SAMPLE_FMT_S16: {
int16_t *sp = table;
*sp++ = (int16_t)d;
table = sp;
continue; }
case AV_SAMPLE_FMT_S32: {
int32_t *ip = table;
*ip++ = (int32_t)d;
table = ip;
continue; }
default:
av_assert0(0);
}
}
}
#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
#define PHASER_PLANAR(name, type) \
static void phaser_## name ##p(AudioPhaserContext *p, \
uint8_t * const *src, uint8_t **dst, \
int nb_samples, int channels) \
{ \
int i, c, delay_pos, modulation_pos; \
\
for (c = 0; c < channels; c++) { \
type *s = (type *)src[c]; \
type *d = (type *)dst[c]; \
double *buffer = p->delay_buffer + \
c * p->delay_buffer_length; \
\
delay_pos = p->delay_pos; \
modulation_pos = p->modulation_pos; \
\
for (i = 0; i < nb_samples; i++, s++, d++) { \
double v = *s * p->in_gain + buffer[ \
MOD(delay_pos + p->modulation_buffer[ \
modulation_pos], \
p->delay_buffer_length)] * p->decay; \
\
modulation_pos = MOD(modulation_pos + 1, \
p->modulation_buffer_length); \
delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
buffer[delay_pos] = v; \
\
*d = v * p->out_gain; \
} \
} \
\
p->delay_pos = delay_pos; \
p->modulation_pos = modulation_pos; \
}
#define PHASER(name, type) \
static void phaser_## name (AudioPhaserContext *p, \
uint8_t * const *src, uint8_t **dst, \
int nb_samples, int channels) \
{ \
int i, c, delay_pos, modulation_pos; \
type *s = (type *)src[0]; \
type *d = (type *)dst[0]; \
double *buffer = p->delay_buffer; \
\
delay_pos = p->delay_pos; \
modulation_pos = p->modulation_pos; \
\
for (i = 0; i < nb_samples; i++) { \
int pos = MOD(delay_pos + p->modulation_buffer[modulation_pos], \
p->delay_buffer_length) * channels; \
int npos; \
\
delay_pos = MOD(delay_pos + 1, p->delay_buffer_length); \
npos = delay_pos * channels; \
for (c = 0; c < channels; c++, s++, d++) { \
double v = *s * p->in_gain + buffer[pos + c] * p->decay; \
\
buffer[npos + c] = v; \
\
*d = v * p->out_gain; \
} \
\
modulation_pos = MOD(modulation_pos + 1, \
p->modulation_buffer_length); \
} \
\
p->delay_pos = delay_pos; \
p->modulation_pos = modulation_pos; \
}
PHASER_PLANAR(dbl, double)
PHASER_PLANAR(flt, float)
PHASER_PLANAR(s16, int16_t)
PHASER_PLANAR(s32, int32_t)
PHASER(dbl, double)
PHASER(flt, float)
PHASER(s16, int16_t)
PHASER(s32, int32_t)
static int config_output(AVFilterLink *outlink)
{
AudioPhaserContext *p = outlink->src->priv;
AVFilterLink *inlink = outlink->src->inputs[0];
p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
p->modulation_buffer = av_malloc(p->modulation_buffer_length * sizeof(*p->modulation_buffer));
if (!p->modulation_buffer || !p->delay_buffer)
return AVERROR(ENOMEM);
generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
p->modulation_buffer, p->modulation_buffer_length,
1., p->delay_buffer_length, M_PI / 2.0);
p->delay_pos = p->modulation_pos = 0;
switch (inlink->format) {
case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
default: av_assert0(0);
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
{
AudioPhaserContext *p = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
AVFrame *outbuf;
if (av_frame_is_writable(inbuf)) {
outbuf = inbuf;
} else {
outbuf = ff_get_audio_buffer(inlink, inbuf->nb_samples);
if (!outbuf)
return AVERROR(ENOMEM);
av_frame_copy_props(outbuf, inbuf);
}
p->phaser(p, inbuf->extended_data, outbuf->extended_data,
outbuf->nb_samples, av_frame_get_channels(outbuf));
if (inbuf != outbuf)
av_frame_free(&inbuf);
return ff_filter_frame(outlink, outbuf);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioPhaserContext *p = ctx->priv;
av_freep(&p->delay_buffer);
av_freep(&p->modulation_buffer);
}
static const AVFilterPad aphaser_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad aphaser_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
{ NULL }
};
static const char *const shorthand[] = { "in_gain", "out_gain", "delay", "decay", "speed", "type", NULL };
AVFilter avfilter_af_aphaser = {
.name = "aphaser",
.description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
.query_formats = query_formats,
.priv_size = sizeof(AudioPhaserContext),
.init = init,
.uninit = uninit,
.inputs = aphaser_inputs,
.outputs = aphaser_outputs,
.priv_class = &aphaser_class,
.shorthand = shorthand,
};

1
libavfilter/allfilters.c
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@ -54,6 +54,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(ANULL, anull, af);
REGISTER_FILTER(APAD, apad, af);
REGISTER_FILTER(APERMS, aperms, af);
REGISTER_FILTER(APHASER, aphaser, af);
REGISTER_FILTER(ARESAMPLE, aresample, af);
REGISTER_FILTER(ASELECT, aselect, af);
REGISTER_FILTER(ASENDCMD, asendcmd, af);

4
libavfilter/version.h
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@ -29,8 +29,8 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 3
#define LIBAVFILTER_VERSION_MINOR 48
#define LIBAVFILTER_VERSION_MICRO 105
#define LIBAVFILTER_VERSION_MINOR 49
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \