mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
302 lines
12 KiB
C
302 lines
12 KiB
C
/*
|
|
* 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;
|
|
}
|
|
|
|
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
|
|
};
|
|
int ret;
|
|
|
|
layouts = ff_all_channel_layouts();
|
|
if (!layouts)
|
|
return AVERROR(ENOMEM);
|
|
ret = ff_set_common_channel_layouts(ctx, layouts);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
formats = ff_make_format_list(sample_fmts);
|
|
if (!formats)
|
|
return AVERROR(ENOMEM);
|
|
ret = ff_set_common_formats(ctx, formats);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
formats = ff_all_samplerates();
|
|
if (!formats)
|
|
return AVERROR(ENOMEM);
|
|
return ff_set_common_samplerates(ctx, formats);
|
|
}
|
|
|
|
#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->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(inlink, inbuf->nb_samples);
|
|
if (!outbuf)
|
|
return AVERROR(ENOMEM);
|
|
av_frame_copy_props(outbuf, inbuf);
|
|
}
|
|
|
|
s->phaser(s, 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 *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,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
static const AVFilterPad aphaser_outputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.config_props = config_output,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
AVFilter ff_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,
|
|
};
|