/* * 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; } }