Fixed rounding of floats for working with negative numbers.

dev
Holger Wirtz 5 years ago
parent 29268ab1fa
commit 5c987723d6
  1. 2
      config.h
  2. 38
      effect_modulated_delay.cpp

@ -57,7 +57,7 @@
#define USE_XFADE_DATA 1
// CHORUS parameters
#define INTERPOLATION_WINDOW_SIZE 21 // use only odd numbers!!!
#define INTERPOLATE_MODE 11
//#define INTERPOLATE_MODE 11
#define CHORUS_WAVEFORM WAVEFORM_SINE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE
#define CHORUS_DELAY_LENGTH_SAMPLES (15*AUDIO_BLOCK_SAMPLES) // one AUDIO_BLOCK_SAMPLES = 2.902ms; you need doubled length, e.g. delay point is 20ms, so you need up to 40ms delay!

@ -87,7 +87,8 @@ void AudioEffectModulatedDelay::update(void)
if (block && modulation)
{
float _tmp;
float mod_number;
float mod_fraction;
for (uint16_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++)
{
@ -99,11 +100,12 @@ void AudioEffectModulatedDelay::update(void)
// Calculate modulation index as a float, for interpolation later.
// The index is located around the half of the delay length multiplied by the current amount of the modulator
mod_idx = float(*mp) / SHRT_MAX * float(_delay_length >> 1);
mod_fraction = modff(mod_idx, &mod_number);
#ifdef INTERPOLATE_MODE
// Generate a an array with the size of INTERPOLATION_WINDOW_SIZE of x/y values around mod_idx for interpolation
uint8_t c;
int16_t c_mod_idx = _circ_idx - int(mod_idx + 0.5); // This is the pointer to the value in the circular buffer at the current modulation index
int16_t c_mod_idx = _circ_idx - int(round(mod_idx)); // This is the pointer to the value in the circular buffer at the current modulation index
for (j = ~(INTERPOLATION_WINDOW_SIZE >> 1) | 0x01, c = 0; j <= INTERPOLATION_WINDOW_SIZE >> 1; j++, c++) // only another way to say: from -INTERPOLATION_WINDOW_SIZE/2 to INTERPOLATION_WINDOW_SIZE/2
{
int16_t jc_mod_idx = (c_mod_idx + j) % _delay_length; // The modulation index pointer plus the value of the current window pointer
@ -113,39 +115,23 @@ void AudioEffectModulatedDelay::update(void)
y[c] = float(_delayline[jc_mod_idx]);
x[c] = float(j);
}
*bp = int(s.value(modff(mod_idx, &_tmp)) + 0.5);
*bp = int(round(s.value(mod_fraction)));
#else
// Simple interpolation
int16_t c_mod_idx = (_circ_idx - int(mod_idx + 0.5)) % _delay_length;
float fraction = modff(mod_idx, &_tmp);
int16_t c_mod_idx = (_circ_idx - int(round(mod_idx))) % _delay_length;
float value1, value2;
if (c_mod_idx < 0)
{
if (fraction < 0.5)
{
value1 = _delayline[_delay_length + c_mod_idx - 1];
value2 = _delayline[_delay_length + c_mod_idx];
}
else
{
value1 = _delayline[_delay_length + c_mod_idx];
value2 = _delayline[_delay_length + c_mod_idx + 1];
}
value1 = _delayline[_delay_length + c_mod_idx];
value2 = _delayline[_delay_length + c_mod_idx - 1];
}
else
{
if (fraction < 0.5)
{
value1 = _delayline[c_mod_idx - 1];
value2 = _delayline[c_mod_idx];
}
else
{
value1 = _delayline[c_mod_idx];
value2 = _delayline[c_mod_idx + 1];
}
value1 = _delayline[c_mod_idx];
value2 = _delayline[c_mod_idx - 1];
}
*bp = int ((fraction * value1) + ((1.0 - fraction) * value2) + 0.5);
*bp = mod_fraction * value1 + (1.0 - mod_fraction) * value2;
#endif
bp++; // next audio data

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