RetroArch/libretro-common/audio/dsp_filters/chorus.c

156 lines
5.2 KiB
C

/* Copyright (C) 2010-2020 The RetroArch team
*
* ---------------------------------------------------------------------------------------
* The following license statement only applies to this file (chorus.c).
* ---------------------------------------------------------------------------------------
*
* Permission is hereby granted, free of charge,
* to any person obtaining a copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
* INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
* WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <retro_miscellaneous.h>
#include <libretro_dspfilter.h>
#define CHORUS_MAX_DELAY 4096
#define CHORUS_DELAY_MASK (CHORUS_MAX_DELAY - 1)
struct chorus_data
{
float old[2][CHORUS_MAX_DELAY];
float delay;
float depth;
float input_rate;
float mix_dry;
float mix_wet;
unsigned old_ptr;
unsigned lfo_ptr;
unsigned lfo_period;
};
static void chorus_free(void *data)
{
if (data)
free(data);
}
static void chorus_process(void *data, struct dspfilter_output *output,
const struct dspfilter_input *input)
{
unsigned i;
float *out = NULL;
struct chorus_data *ch = (struct chorus_data*)data;
output->samples = input->samples;
output->frames = input->frames;
out = output->samples;
for (i = 0; i < input->frames; i++, out += 2)
{
unsigned delay_int;
float delay_frac, l_a, l_b, r_a, r_b;
float chorus_l, chorus_r;
float in[2] = { out[0], out[1] };
float delay = ch->delay + ch->depth * sin((2.0 * M_PI * ch->lfo_ptr++) / ch->lfo_period);
delay *= ch->input_rate;
if (ch->lfo_ptr >= ch->lfo_period)
ch->lfo_ptr = 0;
delay_int = (unsigned)delay;
if (delay_int >= CHORUS_MAX_DELAY - 1)
delay_int = CHORUS_MAX_DELAY - 2;
delay_frac = delay - delay_int;
ch->old[0][ch->old_ptr] = in[0];
ch->old[1][ch->old_ptr] = in[1];
l_a = ch->old[0][(ch->old_ptr - delay_int - 0) & CHORUS_DELAY_MASK];
l_b = ch->old[0][(ch->old_ptr - delay_int - 1) & CHORUS_DELAY_MASK];
r_a = ch->old[1][(ch->old_ptr - delay_int - 0) & CHORUS_DELAY_MASK];
r_b = ch->old[1][(ch->old_ptr - delay_int - 1) & CHORUS_DELAY_MASK];
/* Lerp introduces aliasing of the chorus component,
* but doing full polyphase here is probably overkill. */
chorus_l = l_a * (1.0f - delay_frac) + l_b * delay_frac;
chorus_r = r_a * (1.0f - delay_frac) + r_b * delay_frac;
out[0] = ch->mix_dry * in[0] + ch->mix_wet * chorus_l;
out[1] = ch->mix_dry * in[1] + ch->mix_wet * chorus_r;
ch->old_ptr = (ch->old_ptr + 1) & CHORUS_DELAY_MASK;
}
}
static void *chorus_init(const struct dspfilter_info *info,
const struct dspfilter_config *config, void *userdata)
{
float delay, depth, lfo_freq, drywet;
struct chorus_data *ch = (struct chorus_data*)calloc(1, sizeof(*ch));
if (!ch)
return NULL;
config->get_float(userdata, "delay_ms", &delay, 25.0f);
config->get_float(userdata, "depth_ms", &depth, 1.0f);
config->get_float(userdata, "lfo_freq", &lfo_freq, 0.5f);
config->get_float(userdata, "drywet", &drywet, 0.8f);
delay /= 1000.0f;
depth /= 1000.0f;
if (depth > delay)
depth = delay;
if (drywet < 0.0f)
drywet = 0.0f;
else if (drywet > 1.0f)
drywet = 1.0f;
ch->mix_dry = 1.0f - 0.5f * drywet;
ch->mix_wet = 0.5f * drywet;
ch->delay = delay;
ch->depth = depth;
ch->lfo_period = (1.0f / lfo_freq) * info->input_rate;
ch->input_rate = info->input_rate;
if (!ch->lfo_period)
ch->lfo_period = 1;
return ch;
}
static const struct dspfilter_implementation chorus_plug = {
chorus_init,
chorus_process,
chorus_free,
DSPFILTER_API_VERSION,
"Chorus",
"chorus",
};
#ifdef HAVE_FILTERS_BUILTIN
#define dspfilter_get_implementation chorus_dspfilter_get_implementation
#endif
const struct dspfilter_implementation *
dspfilter_get_implementation(dspfilter_simd_mask_t mask) { return &chorus_plug; }
#undef dspfilter_get_implementation