wirtz
/
MicroMDAEPiano
mirror of https://codeberg.org/dcoredump/MicroMDAEPiano.git
1
0
Fork 0
Code Issues Releases Wiki Activity

Replaced own filter functions with CMSIS library function calls.

Browse Source
master
Holger Wirtz 5 years ago
parent 807d152020
commit cd891c4f98
3 changed files with 70 additions and 77 deletions
Show all changes Ignore whitespace when comparing lines Ignore changes in amount of whitespace Ignore changes in whitespace at EOL
Show Stats Download Patch File Download Diff File
  1. 5
      config.h
  2. 131
      effect_modulated_delay.cpp
  3. 11
      effect_modulated_delay.h

5
config.h
Unescape View File

@ -56,9 +56,8 @@
#define REDUCE_LOUDNESS 0 #define REDUCE_LOUDNESS 0
#define USE_XFADE_DATA 1 #define USE_XFADE_DATA 1
// CHORUS parameters // CHORUS parameters
#define INTERPOLATION_WINDOW_SIZE 7 // use only odd numbers!!! #define INTERPOLATION_WINDOW_SIZE 5 // use only odd numbers!!!
#define INTERPOLATE_MODE 11 //#define INTERPOLATE_MODE 11
#define FILTER_IIR_ORDER_3 1
#define CHORUS_WAVEFORM WAVEFORM_TRIANGLE // WAVEFORM_SINE WAVEFORM_TRIANGLE WAVEFORM_SAWTOOTH WAVEFORM_SAWTOOTH_REVERSE #define CHORUS_WAVEFORM WAVEFORM_TRIANGLE // 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! #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!

131
effect_modulated_delay.cpp
Unescape View File

@ -58,6 +58,12 @@ boolean AudioEffectModulatedDelay::begin(short *delayline, int d_length)
_delayline = delayline; _delayline = delayline;
_delay_length = _max_delay_length = d_length; _delay_length = _max_delay_length = d_length;
// init filter
filter.numStages = 1;
filter.pState = filter_state;
filter.pCoeffs = filter_coeffs;
calcModFilterCoeff(5000.0, 0.0, 5.0);
return (true); return (true);
} }
@ -66,31 +72,33 @@ void AudioEffectModulatedDelay::update(void)
audio_block_t *block; audio_block_t *block;
audio_block_t *modulation; audio_block_t *modulation;
int16_t *bp;
int16_t *mp;
float mod_idx;
if (_delayline == NULL) if (_delayline == NULL)
return; return;
block = receiveWritable(0); block = receiveWritable(0);
modulation = receiveReadOnly(1); modulation = receiveReadOnly(1);
#ifdef INTERPOLATE_MODE
int8_t j;
float x[INTERPOLATION_WINDOW_SIZE];
float y[INTERPOLATION_WINDOW_SIZE];
Spline s(x, y, INTERPOLATION_WINDOW_SIZE, INTERPOLATE_MODE);
#endif
bp = block->data;
mp = modulation->data;
if (block && modulation) if (block && modulation)
{ {
int16_t *bp;
float *mp;
float mod_idx;
float mod_number; float mod_number;
float mod_fraction; float mod_fraction;
#ifdef INTERPOLATE_MODE
int8_t j;
float x[INTERPOLATION_WINDOW_SIZE];
float y[INTERPOLATION_WINDOW_SIZE];
Spline s(x, y, INTERPOLATION_WINDOW_SIZE, INTERPOLATE_MODE);
#endif
// (Filter implementation: https://web.fhnw.ch/technik/projekte/eit/Fruehling2016/MuelZum/html/parametric__equalizer__example_8c_source.html)
arm_q15_to_float(modulation->data, modulation_f32, AUDIO_BLOCK_SAMPLES);
arm_biquad_cascade_df1_f32(&filter, modulation_f32, modulation_f32, AUDIO_BLOCK_SAMPLES);
bp = block->data;
mp = modulation_f32;
for (uint16_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++) for (uint16_t i = 0; i < AUDIO_BLOCK_SAMPLES; i++)
{ {
// write data into circular buffer // write data into circular buffer
@ -100,56 +108,7 @@ void AudioEffectModulatedDelay::update(void)
// Calculate modulation index as a float, for interpolation later. // 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 // The index is located around the half of the delay length multiplied by the current amount of the modulator
/************************************************************************************************************** mod_idx = *mp * float(_delay_length >> 1);
Use an IIR filter on the modulation for avoiding aliasing (http://www-users.cs.york.ac.uk/~fisher/mkfilter/) */
#if defined(FILTER_IIR_ORDER_1)
/* Digital filter designed by mkfilter/mkshape/gencode A.J. Fisher
Command line: /www/usr/fisher/helpers/mkfilter -Bu -Lp -o 1 -a 1.1333499558e-01 0.0000000000e+00 -l
Parameters: filtertype = Butterworth
passtype = Lowpass
ripple =
order = 1
samplerate = 44117.64706
corner1 = 5000
corner2 =
adzero =
logmin =
**************************************************************************************************************/
#define NZEROS 1
#define NPOLES 1
#define GAIN 3.688918967e+00
static float xv[NZEROS + 1], yv[NPOLES + 1];
xv[0] = xv[1];
xv[1] = (float(*mp) / SHRT_MAX) / GAIN;
yv[0] = yv[1];
yv[1] = (xv[0] + xv[1]) + (0.4578357460 * yv[0]);
//new_value = yv[1];
mod_idx = yv[1] * float(_delay_length >> 1);
#elif defined(FILTER_IIR_ORDER_3)
/* Digital filter designed by mkfilter/mkshape/gencode A.J. Fisher
Command line: /www/usr/fisher/helpers/mkfilter -Bu -Lp -o 3 -a 1.1333333333e-01 0.0000000000e+00 -l
raw alpha1 = 0.1133333333
raw alpha2 = 0.1133333333
warped alpha1 = 0.1183783855
warped alpha2 = 0.1183783855
gain at dc : mag = 4.028005941e+01 phase = 0.0000000000 pi
gain at centre: mag = 2.848230315e+01 phase = -0.7500000000 pi
gain at hf : mag = 0.000000000e+00
**************************************************************************************************************/
#define NZEROS 3
#define NPOLES 3
#define GAIN 4.028005941e+01
static float xv[NZEROS + 1], yv[NPOLES + 1];
xv[0] = xv[1]; xv[1] = xv[2]; xv[2] = xv[3];
xv[3] = (float(*mp) / SHRT_MAX) / GAIN;
yv[0] = yv[1]; yv[1] = yv[2]; yv[2] = yv[3];
yv[3] = (xv[0] + xv[3]) + 3 * (xv[1] + xv[2])
+ ( 0.2323461955 * yv[0]) + ( -1.0299524229 * yv[1])
+ ( 1.5989967885 * yv[2]);
//new_value = yv[3];
mod_idx = yv[3] * float(_delay_length >> 1);
/************************************************************************/
#endif
mod_fraction = modff(mod_idx, &mod_number); mod_fraction = modff(mod_idx, &mod_number);
#ifdef INTERPOLATE_MODE #ifdef INTERPOLATE_MODE
@ -173,13 +132,13 @@ void AudioEffectModulatedDelay::update(void)
if (c_mod_idx < 0) if (c_mod_idx < 0)
{ {
value1 = _delayline[_delay_length + c_mod_idx]; value1 = _delayline[_delay_length + c_mod_idx - 1];
value2 = _delayline[_delay_length + c_mod_idx - 1]; value2 = _delayline[_delay_length + c_mod_idx];
} }
else else
{ {
value1 = _delayline[c_mod_idx]; value1 = _delayline[c_mod_idx - 1];
value2 = _delayline[c_mod_idx - 1]; value2 = _delayline[c_mod_idx];
} }
*bp = mod_fraction * value1 + (1.0 - mod_fraction) * value2; *bp = mod_fraction * value1 + (1.0 - mod_fraction) * value2;
#endif #endif
@ -190,17 +149,47 @@ void AudioEffectModulatedDelay::update(void)
} }
} }
if (modulation)
release(modulation);
if (block) if (block)
{ {
transmit(block, 0); transmit(block, 0);
release(block); release(block);
} }
if (modulation)
release(modulation);
} }
void AudioEffectModulatedDelay::setDelay(float milliseconds) void AudioEffectModulatedDelay::setDelay(float milliseconds)
{ {
_delay_length = min(AUDIO_SAMPLE_RATE * milliseconds / 500, _max_delay_length); _delay_length = min(AUDIO_SAMPLE_RATE * milliseconds / 500, _max_delay_length);
} }
void AudioEffectModulatedDelay::calcModFilterCoeff(float32_t cFrq, float32_t gain, float32_t width)
{
/* Calculate intermediate values */
float32_t A = sqrt(pow(10, gain / 20.0f));
float32_t w0 = 2.0f * PI * cFrq / ((float32_t)AUDIO_SAMPLE_RATE_EXACT);
float32_t cosw0 = cos(w0);
float32_t sinw0 = sin(w0);
float32_t alpha = sinw0 / (2.0f * width);
/* Calculate coefficients */
float32_t b0 = 1.0f + alpha * A;
float32_t b1 = -2.0f * cosw0;
float32_t b2 = 1.0f - alpha * A;
float32_t a0 = 1.0f + alpha / A;
float32_t a1 = -2.0f * cosw0;
float32_t a2 = 1.0f - alpha / A;
/* Normalize so a0 = 1 */
filter_coeffs[0] = b0 / a0;
filter_coeffs[1] = b1 / a0;
filter_coeffs[2] = b2 / a0;
filter_coeffs[3] = -a1 / a0;
filter_coeffs[4] = -a2 / a0;
}
void AudioEffectModulatedDelay::setModFilter(float cFrq, float gain, float width)
{
calcModFilterCoeff(cFrq, gain, width);
}

11
effect_modulated_delay.h
Unescape View File

@ -34,9 +34,6 @@
// 140219 - correct storage class (not static) // 140219 - correct storage class (not static)
// 190527 - added modulation input handling (by Holger Wirtz) // 190527 - added modulation input handling (by Holger Wirtz)
#define MODF(n,i,f) ((i) = (int)(n), (f) = (n) - (double)(i))
class AudioEffectModulatedDelay : class AudioEffectModulatedDelay :
public AudioStream public AudioStream
{ {
@ -48,13 +45,21 @@ class AudioEffectModulatedDelay :
boolean begin(short *delayline, int delay_length); boolean begin(short *delayline, int delay_length);
virtual void update(void); virtual void update(void);
virtual void setDelay(float milliseconds); virtual void setDelay(float milliseconds);
virtual void setModFilter(float cFrq, float gain, float width);
private: private:
virtual void calcModFilterCoeff(float32_t cFrq, float32_t gain, float32_t width);
audio_block_t *inputQueueArray[2]; audio_block_t *inputQueueArray[2];
int16_t *_delayline; int16_t *_delayline;
uint16_t _circ_idx; uint16_t _circ_idx;
uint16_t _max_delay_length; uint16_t _max_delay_length;
uint16_t _delay_length; uint16_t _delay_length;
// filter data
arm_biquad_casd_df1_inst_f32 filter;
float32_t modulation_f32[AUDIO_BLOCK_SAMPLES];
float32_t filter_coeffs[5];
float32_t filter_state[4];
}; };
#endif #endif

Write Preview
Loading…
Cancel
Save