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ardour/libs/fluidsynth/src/fluid_chorus.c

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/*
* August 24, 1998
* Copyright (C) 1998 Juergen Mueller And Sundry Contributors
* This source code is freely redistributable and may be used for
* any purpose. This copyright notice must be maintained.
* Juergen Mueller And Sundry Contributors are not responsible for
* the consequences of using this software.
*/
/*
CHANGES
- Adapted for fluidsynth, Peter Hanappe, March 2002
- Variable delay line implementation using bandlimited
interpolation, code reorganization: Markus Nentwig May 2002
*/
/*
* Chorus effect.
*
* Flow diagram scheme for n delays ( 1 <= n <= MAX_CHORUS ):
*
* * gain-in ___
* ibuff -----+--------------------------------------------->| |
* | _________ | |
* | | | * level 1 | |
* +---->| delay 1 |----------------------------->| |
* | |_________| | |
* | /|\ | |
* : | | |
* : +-----------------+ +--------------+ | + |
* : | Delay control 1 |<--| mod. speed 1 | | |
* : +-----------------+ +--------------+ | |
* | _________ | |
* | | | * level n | |
* +---->| delay n |----------------------------->| |
* |_________| | |
* /|\ |___|
* | |
* +-----------------+ +--------------+ | * gain-out
* | Delay control n |<--| mod. speed n | |
* +-----------------+ +--------------+ +----->obuff
*
*
* The delay i is controlled by a sine or triangle modulation i ( 1 <= i <= n).
*
* The delay of each block is modulated between 0..depth ms
*
*/
/* Variable delay line implementation
* ==================================
*
* The modulated delay needs the value of the delayed signal between
* samples. A lowpass filter is used to obtain intermediate values
* between samples (bandlimited interpolation). The sample pulse
* train is convoluted with the impulse response of the low pass
* filter (sinc function). To make it work with a small number of
* samples, the sinc function is windowed (Hamming window).
*
*/
#include "fluid_chorus.h"
#include "fluid_sys.h"
#define MAX_CHORUS 99
#define MAX_DELAY 100
#define MAX_DEPTH 10
#define MIN_SPEED_HZ 0.29
#define MAX_SPEED_HZ 5
/* Length of one delay line in samples:
* Set through MAX_SAMPLES_LN2.
* For example:
* MAX_SAMPLES_LN2=12
* => MAX_SAMPLES=pow(2,12)=4096
* => MAX_SAMPLES_ANDMASK=4095
*/
#define MAX_SAMPLES_LN2 12
#define MAX_SAMPLES (1 << (MAX_SAMPLES_LN2-1))
#define MAX_SAMPLES_ANDMASK (MAX_SAMPLES-1)
/* Interpolate how many steps between samples? Must be power of two
For example: 8 => use a resolution of 256 steps between any two
samples
*/
#define INTERPOLATION_SUBSAMPLES_LN2 8
#define INTERPOLATION_SUBSAMPLES (1 << (INTERPOLATION_SUBSAMPLES_LN2-1))
#define INTERPOLATION_SUBSAMPLES_ANDMASK (INTERPOLATION_SUBSAMPLES-1)
/* Use how many samples for interpolation? Must be odd. '7' sounds
relatively clean, when listening to the modulated delay signal
alone. For a demo on aliasing try '1' With '3', the aliasing is
still quite pronounced for some input frequencies
*/
#define INTERPOLATION_SAMPLES 5
/* Private data for SKEL file */
struct _fluid_chorus_t {
int type;
fluid_real_t depth_ms;
fluid_real_t level;
fluid_real_t speed_Hz;
int number_blocks;
fluid_real_t *chorusbuf;
int counter;
long phase[MAX_CHORUS];
long modulation_period_samples;
int *lookup_tab;
fluid_real_t sample_rate;
/* sinc lookup table */
fluid_real_t sinc_table[INTERPOLATION_SAMPLES][INTERPOLATION_SUBSAMPLES];
};
static void fluid_chorus_triangle(int *buf, int len, int depth);
static void fluid_chorus_sine(int *buf, int len, int depth);
fluid_chorus_t*
new_fluid_chorus(fluid_real_t sample_rate)
{
int i; int ii;
fluid_chorus_t* chorus;
chorus = FLUID_NEW(fluid_chorus_t);
if (chorus == NULL) {
fluid_log(FLUID_PANIC, "chorus: Out of memory");
return NULL;
}
FLUID_MEMSET(chorus, 0, sizeof(fluid_chorus_t));
chorus->sample_rate = sample_rate;
/* Lookup table for the SI function (impulse response of an ideal low pass) */
/* i: Offset in terms of whole samples */
for (i = 0; i < INTERPOLATION_SAMPLES; i++){
/* ii: Offset in terms of fractional samples ('subsamples') */
for (ii = 0; ii < INTERPOLATION_SUBSAMPLES; ii++){
/* Move the origin into the center of the table */
double i_shifted = ((double) i- ((double) INTERPOLATION_SAMPLES) / 2.
+ (double) ii / (double) INTERPOLATION_SUBSAMPLES);
if (fabs(i_shifted) < 0.000001) {
/* sinc(0) cannot be calculated straightforward (limit needed
for 0/0) */
chorus->sinc_table[i][ii] = (fluid_real_t)1.;
} else {
chorus->sinc_table[i][ii] = (fluid_real_t)sin(i_shifted * M_PI) / (M_PI * i_shifted);
/* Hamming window */
chorus->sinc_table[i][ii] *= (fluid_real_t)0.5 * (1.0 + cos(2.0 * M_PI * i_shifted / (fluid_real_t)INTERPOLATION_SAMPLES));
};
};
};
/* allocate lookup tables */
chorus->lookup_tab = FLUID_ARRAY(int, (int) (chorus->sample_rate / MIN_SPEED_HZ));
if (chorus->lookup_tab == NULL) {
fluid_log(FLUID_PANIC, "chorus: Out of memory");
goto error_recovery;
}
/* allocate sample buffer */
chorus->chorusbuf = FLUID_ARRAY(fluid_real_t, MAX_SAMPLES);
if (chorus->chorusbuf == NULL) {
fluid_log(FLUID_PANIC, "chorus: Out of memory");
goto error_recovery;
}
if (fluid_chorus_init(chorus) != FLUID_OK){
goto error_recovery;
};
return chorus;
error_recovery:
delete_fluid_chorus(chorus);
return NULL;
}
void
delete_fluid_chorus(fluid_chorus_t* chorus)
{
if (chorus == NULL) {
return;
}
if (chorus->chorusbuf != NULL) {
FLUID_FREE(chorus->chorusbuf);
}
if (chorus->lookup_tab != NULL) {
FLUID_FREE(chorus->lookup_tab);
}
FLUID_FREE(chorus);
}
int
fluid_chorus_init(fluid_chorus_t* chorus)
{
int i;
for (i = 0; i < MAX_SAMPLES; i++) {
chorus->chorusbuf[i] = 0.0;
}
/* initialize the chorus with the default settings */
fluid_chorus_set (chorus, FLUID_CHORUS_SET_ALL, FLUID_CHORUS_DEFAULT_N,
FLUID_CHORUS_DEFAULT_LEVEL, FLUID_CHORUS_DEFAULT_SPEED,
FLUID_CHORUS_DEFAULT_DEPTH, FLUID_CHORUS_MOD_SINE);
return FLUID_OK;
}
void
fluid_chorus_reset(fluid_chorus_t* chorus)
{
fluid_chorus_init(chorus);
}
/**
* Set one or more chorus parameters.
* @param chorus Chorus instance
* @param set Flags indicating which chorus parameters to set (#fluid_chorus_set_t)
* @param nr Chorus voice count (0-99, CPU time consumption proportional to
* this value)
* @param level Chorus level (0.0-10.0)
* @param speed Chorus speed in Hz (0.29-5.0)
* @param depth_ms Chorus depth (max value depends on synth sample rate,
* 0.0-21.0 is safe for sample rate values up to 96KHz)
* @param type Chorus waveform type (#fluid_chorus_mod)
*/
void
fluid_chorus_set(fluid_chorus_t* chorus, int set, int nr, float level,
float speed, float depth_ms, int type)
{
int modulation_depth_samples;
int i;
if (set & FLUID_CHORUS_SET_NR) chorus->number_blocks = nr;
if (set & FLUID_CHORUS_SET_LEVEL) chorus->level = level;
if (set & FLUID_CHORUS_SET_SPEED) chorus->speed_Hz = speed;
if (set & FLUID_CHORUS_SET_DEPTH) chorus->depth_ms = depth_ms;
if (set & FLUID_CHORUS_SET_TYPE) chorus->type = type;
if (chorus->number_blocks < 0) {
fluid_log(FLUID_WARN, "chorus: number blocks must be >=0! Setting value to 0.");
chorus->number_blocks = 0;
} else if (chorus->number_blocks > MAX_CHORUS) {
fluid_log(FLUID_WARN, "chorus: number blocks larger than max. allowed! Setting value to %d.",
MAX_CHORUS);
chorus->number_blocks = MAX_CHORUS;
}
if (chorus->speed_Hz < MIN_SPEED_HZ) {
fluid_log(FLUID_WARN, "chorus: speed is too low (min %f)! Setting value to min.",
(double) MIN_SPEED_HZ);
chorus->speed_Hz = MIN_SPEED_HZ;
} else if (chorus->speed_Hz > MAX_SPEED_HZ) {
fluid_log(FLUID_WARN, "chorus: speed must be below %f Hz! Setting value to max.",
(double) MAX_SPEED_HZ);
chorus->speed_Hz = MAX_SPEED_HZ;
}
if (chorus->depth_ms < 0.0) {
fluid_log(FLUID_WARN, "chorus: depth must be positive! Setting value to 0.");
chorus->depth_ms = 0.0;
}
/* Depth: Check for too high value through modulation_depth_samples. */
if (chorus->level < 0.0) {
fluid_log(FLUID_WARN, "chorus: level must be positive! Setting value to 0.");
chorus->level = 0.0;
} else if (chorus->level > 10) {
fluid_log(FLUID_WARN, "chorus: level must be < 10. A reasonable level is << 1! "
"Setting it to 0.1.");
chorus->level = 0.1;
}
/* The modulating LFO goes through a full period every x samples: */
chorus->modulation_period_samples = chorus->sample_rate / chorus->speed_Hz;
/* The variation in delay time is x: */
modulation_depth_samples = (int)
(chorus->depth_ms / 1000.0 /* convert modulation depth in ms to s*/
* chorus->sample_rate);
if (modulation_depth_samples > MAX_SAMPLES) {
fluid_log(FLUID_WARN, "chorus: Too high depth. Setting it to max (%d).", MAX_SAMPLES);
modulation_depth_samples = MAX_SAMPLES;
}
/* initialize LFO table */
if (chorus->type == FLUID_CHORUS_MOD_SINE) {
fluid_chorus_sine(chorus->lookup_tab, chorus->modulation_period_samples,
modulation_depth_samples);
} else if (chorus->type == FLUID_CHORUS_MOD_TRIANGLE) {
fluid_chorus_triangle(chorus->lookup_tab, chorus->modulation_period_samples,
modulation_depth_samples);
} else {
fluid_log(FLUID_WARN, "chorus: Unknown modulation type. Using sinewave.");
chorus->type = FLUID_CHORUS_MOD_SINE;
fluid_chorus_sine(chorus->lookup_tab, chorus->modulation_period_samples,
modulation_depth_samples);
}
for (i = 0; i < chorus->number_blocks; i++) {
/* Set the phase of the chorus blocks equally spaced */
chorus->phase[i] = (int) ((double) chorus->modulation_period_samples
* (double) i / (double) chorus->number_blocks);
}
/* Start of the circular buffer */
chorus->counter = 0;
}
void fluid_chorus_processmix(fluid_chorus_t* chorus, fluid_real_t *in,
fluid_real_t *left_out, fluid_real_t *right_out)
{
int sample_index;
int i;
fluid_real_t d_in, d_out;
for (sample_index = 0; sample_index < FLUID_BUFSIZE; sample_index++) {
d_in = in[sample_index];
d_out = 0.0f;
# if 0
/* Debug: Listen to the chorus signal only */
left_out[sample_index]=0;
right_out[sample_index]=0;
#endif
/* Write the current sample into the circular buffer */
chorus->chorusbuf[chorus->counter] = d_in;
for (i = 0; i < chorus->number_blocks; i++) {
int ii;
/* Calculate the delay in subsamples for the delay line of chorus block nr. */
/* The value in the lookup table is so, that this expression
* will always be positive. It will always include a number of
* full periods of MAX_SAMPLES*INTERPOLATION_SUBSAMPLES to
* remain positive at all times. */
int pos_subsamples = (INTERPOLATION_SUBSAMPLES * chorus->counter
- chorus->lookup_tab[chorus->phase[i]]);
int pos_samples = pos_subsamples/INTERPOLATION_SUBSAMPLES;
/* modulo divide by INTERPOLATION_SUBSAMPLES */
pos_subsamples &= INTERPOLATION_SUBSAMPLES_ANDMASK;
for (ii = 0; ii < INTERPOLATION_SAMPLES; ii++){
/* Add the delayed signal to the chorus sum d_out Note: The
* delay in the delay line moves backwards for increasing
* delay!*/
/* The & in chorusbuf[...] is equivalent to a division modulo
MAX_SAMPLES, only faster. */
d_out += chorus->chorusbuf[pos_samples & MAX_SAMPLES_ANDMASK]
* chorus->sinc_table[ii][pos_subsamples];
pos_samples--;
};
/* Cycle the phase of the modulating LFO */
chorus->phase[i]++;
chorus->phase[i] %= (chorus->modulation_period_samples);
} /* foreach chorus block */
d_out *= chorus->level;
/* Add the chorus sum d_out to output */
left_out[sample_index] += d_out;
right_out[sample_index] += d_out;
/* Move forward in circular buffer */
chorus->counter++;
chorus->counter %= MAX_SAMPLES;
} /* foreach sample */
}
/* Duplication of code ... (replaces sample data instead of mixing) */
void fluid_chorus_processreplace(fluid_chorus_t* chorus, fluid_real_t *in,
fluid_real_t *left_out, fluid_real_t *right_out)
{
int sample_index;
int i;
fluid_real_t d_in, d_out;
for (sample_index = 0; sample_index < FLUID_BUFSIZE; sample_index++) {
d_in = in[sample_index];
d_out = 0.0f;
# if 0
/* Debug: Listen to the chorus signal only */
left_out[sample_index]=0;
right_out[sample_index]=0;
#endif
/* Write the current sample into the circular buffer */
chorus->chorusbuf[chorus->counter] = d_in;
for (i = 0; i < chorus->number_blocks; i++) {
int ii;
/* Calculate the delay in subsamples for the delay line of chorus block nr. */
/* The value in the lookup table is so, that this expression
* will always be positive. It will always include a number of
* full periods of MAX_SAMPLES*INTERPOLATION_SUBSAMPLES to
* remain positive at all times. */
int pos_subsamples = (INTERPOLATION_SUBSAMPLES * chorus->counter
- chorus->lookup_tab[chorus->phase[i]]);
int pos_samples = pos_subsamples / INTERPOLATION_SUBSAMPLES;
/* modulo divide by INTERPOLATION_SUBSAMPLES */
pos_subsamples &= INTERPOLATION_SUBSAMPLES_ANDMASK;
for (ii = 0; ii < INTERPOLATION_SAMPLES; ii++){
/* Add the delayed signal to the chorus sum d_out Note: The
* delay in the delay line moves backwards for increasing
* delay!*/
/* The & in chorusbuf[...] is equivalent to a division modulo
MAX_SAMPLES, only faster. */
d_out += chorus->chorusbuf[pos_samples & MAX_SAMPLES_ANDMASK]
* chorus->sinc_table[ii][pos_subsamples];
pos_samples--;
};
/* Cycle the phase of the modulating LFO */
chorus->phase[i]++;
chorus->phase[i] %= (chorus->modulation_period_samples);
} /* foreach chorus block */
d_out *= chorus->level;
/* Store the chorus sum d_out to output */
left_out[sample_index] = d_out;
right_out[sample_index] = d_out;
/* Move forward in circular buffer */
chorus->counter++;
chorus->counter %= MAX_SAMPLES;
} /* foreach sample */
}
/* Purpose:
*
* Calculates a modulation waveform (sine) Its value ( modulo
* MAXSAMPLES) varies between 0 and depth*INTERPOLATION_SUBSAMPLES.
* Its period length is len. The waveform data will be used modulo
* MAXSAMPLES only. Since MAXSAMPLES is substracted from the waveform
* a couple of times here, the resulting (current position in
* buffer)-(waveform sample) will always be positive.
*/
static void
fluid_chorus_sine(int *buf, int len, int depth)
{
int i;
double val;
for (i = 0; i < len; i++) {
val = sin((double) i / (double)len * 2.0 * M_PI);
buf[i] = (int) ((1.0 + val) * (double) depth / 2.0 * (double) INTERPOLATION_SUBSAMPLES);
buf[i] -= 3* MAX_SAMPLES * INTERPOLATION_SUBSAMPLES;
// printf("%i %i\n",i,buf[i]);
}
}
/* Purpose:
* Calculates a modulation waveform (triangle)
* See fluid_chorus_sine for comments.
*/
static void
fluid_chorus_triangle(int *buf, int len, int depth)
{
int i=0;
int ii=len-1;
double val;
double val2;
while (i <= ii){
val = i * 2.0 / len * (double)depth * (double) INTERPOLATION_SUBSAMPLES;
val2= (int) (val + 0.5) - 3 * MAX_SAMPLES * INTERPOLATION_SUBSAMPLES;
buf[i++] = (int) val2;
buf[ii--] = (int) val2;
}
}
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