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/*
* 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"
#include "generate_wave_table.h"
typedef struct AudioPhaserContext {
const AVClass *class;
double in_gain, out_gain;
double delay;
double decay;
double speed;
int 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 *s,
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)
{
AudioPhaserContext *s = ctx->priv;
if (s->in_gain > (1 - s->decay * s->decay))
av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
if (s->in_gain / (1 - s->decay) > 1 / s->out_gain)
av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
return 0;
}
#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
#define PHASER_PLANAR(name, type) \
static void phaser_## name ##p(AudioPhaserContext *s, \
uint8_t * const *ssrc, uint8_t **ddst, \
int nb_samples, int channels) \
{ \
int i, c, delay_pos, modulation_pos; \
\
av_assert0(channels > 0); \
for (c = 0; c < channels; c++) { \
type *src = (type *)ssrc[c]; \
type *dst = (type *)ddst[c]; \
double *buffer = s->delay_buffer + \
c * s->delay_buffer_length; \
\
delay_pos = s->delay_pos; \
modulation_pos = s->modulation_pos; \
\
for (i = 0; i < nb_samples; i++, src++, dst++) { \
double v = *src * s->in_gain + buffer[ \
MOD(delay_pos + s->modulation_buffer[ \
modulation_pos], \
s->delay_buffer_length)] * s->decay; \
\
modulation_pos = MOD(modulation_pos + 1, \
s->modulation_buffer_length); \
delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
buffer[delay_pos] = v; \
\
*dst = v * s->out_gain; \
} \
} \
\
s->delay_pos = delay_pos; \
s->modulation_pos = modulation_pos; \
}
#define PHASER(name, type) \
static void phaser_## name (AudioPhaserContext *s, \
uint8_t * const *ssrc, uint8_t **ddst, \
int nb_samples, int channels) \
{ \
int i, c, delay_pos, modulation_pos; \
type *src = (type *)ssrc[0]; \
type *dst = (type *)ddst[0]; \
double *buffer = s->delay_buffer; \
\
delay_pos = s->delay_pos; \
modulation_pos = s->modulation_pos; \
\
for (i = 0; i < nb_samples; i++) { \
int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
s->delay_buffer_length) * channels; \
int npos; \
\
delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
npos = delay_pos * channels; \
for (c = 0; c < channels; c++, src++, dst++) { \
double v = *src * s->in_gain + buffer[pos + c] * s->decay; \
\
buffer[npos + c] = v; \
\
*dst = v * s->out_gain; \
} \
\
modulation_pos = MOD(modulation_pos + 1, \
s->modulation_buffer_length); \
} \
\
s->delay_pos = delay_pos; \
s->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 *s = outlink->src->priv;
AVFilterLink *inlink = outlink->src->inputs[0];
s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
if (s->delay_buffer_length <= 0) {
av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
return AVERROR(EINVAL);
}
s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->ch_layout.nb_channels);
s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer));
if (!s->modulation_buffer || !s->delay_buffer)
return AVERROR(ENOMEM);
ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32,
s->modulation_buffer, s->modulation_buffer_length,
1., s->delay_buffer_length, M_PI / 2.0);
s->delay_pos = s->modulation_pos = 0;
switch (inlink->format) {
case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break;
case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break;
case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break;
case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break;
case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
default: av_assert0(0);
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
{
AudioPhaserContext *s = 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(outlink, inbuf->nb_samples);
if (!outbuf) {
av_frame_free(&inbuf);
return AVERROR(ENOMEM);
}
av_frame_copy_props(outbuf, inbuf);
}
s->phaser(s, inbuf->extended_data, outbuf->extended_data,
outbuf->nb_samples, outbuf->ch_layout.nb_channels);
if (inbuf != outbuf)
av_frame_free(&inbuf);
return ff_filter_frame(outlink, outbuf);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioPhaserContext *s = ctx->priv;
av_freep(&s->delay_buffer);
av_freep(&s->modulation_buffer);
}
static const AVFilterPad aphaser_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
static const AVFilterPad aphaser_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_aphaser = {
.name = "aphaser",
.description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
.priv_size = sizeof(AudioPhaserContext),
.init = init,
.uninit = uninit,
FILTER_INPUTS(aphaser_inputs),
FILTER_OUTPUTS(aphaser_outputs),
FILTER_SAMPLEFMTS(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),
.priv_class = &aphaser_class,
};
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