wirtz
/
OpenAudio_ArduinoLibrary
mirror of https://github.com/chipaudette/OpenAudio_ArduinoLibrary.git
1
0
Fork 0
Code Issues Releases Wiki Activity

Added pureSpectrum filtering option.

Browse Source
pull/13/head
boblark 3 years ago
parent 59e11fd79f
commit 8353f29b10
2 changed files with 142 additions and 85 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. 68
      synth_sine_f32.cpp
  2. 149
      synth_sine_f32.h

68
synth_sine_f32.cpp
Unescape View File

@ -7,52 +7,36 @@
* Purpose: Create sine wave of given amplitude and frequency
*
* License: MIT License. Use at your own risk.
*
*
* Revised per synth_sine_f32.h. 7 Feb 2022 Bob.
*/
#include "synth_sine_f32.h"
#include "utility/dspinst.h"
// 513 values of the sine wave in a float array:
#include "sinTable512_f32.h"
// data_waveforms.c
extern "C" {
extern const int16_t AudioWaveformSine[257];
}
void AudioSynthWaveformSine_F32::update(void) {
audio_block_f32_t *block;
uint32_t i, ph, inc, index, scale;
int32_t val1, val2;
static uint32_t block_length = 0;
void AudioSynthWaveformSine_F32::update(void) {
audio_block_f32_t *blockS;
uint16_t index, i;
float32_t a, b;
if (enabled) {
if (magnitude) {
block = allocate_f32();
if (block) {
block_length = (uint32_t)block->length;
ph = phase_accumulator;
inc = phase_increment;
for (i=0; i < block_length; i++) {
index = ph >> 24;
val1 = AudioWaveformSine[index];
val2 = AudioWaveformSine[index+1];
scale = (ph >> 8) & 0xFFFF;
val2 *= scale;
val1 *= 0x10000 - scale;
#if (defined(KINETISK) || defined(__IMXRT1062__) )
block->data[i] = (float) multiply_32x32_rshift32(val1 + val2, magnitude);
#elif defined(KINETISL)
block->data[i] = (float) ((((val1 + val2) >> 16) * magnitude) >> 16);
#endif
ph += inc;
block->data[i] = block->data[i] / 32767.0f; // scale to float
}
phase_accumulator = ph;
blockS = AudioStream_F32::allocate_f32(); // Output blocks
if (!blockS) return;
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
return;
}
}// end if (magnitude)
phase_accumulator += phase_increment * block_length; // continue sine wave while magnitude==0
} // end if (enabled)
for (i=0; i < blockS->length; i++) {
phaseS += phaseIncrement;
if (phaseS > 512.0f) phaseS -= 512.0f;
index = (uint16_t) phaseS;
float32_t deltaPhase = phaseS - (float32_t)index;
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index];
b = sinTable512_f32[index+1];
blockS->data[i] = magnitude*(a+(b-a)*deltaPhase); // Linear interpolation
}
// For higher frequencies, an optional bandpass filter the output
// This does a pass through for lower frequencies
if(doPureSpectrum)
arm_biquad_cascade_df1_f32(&bq_inst, blockS->data, blockS->data, 128);
AudioStream_F32::transmit(blockS);
AudioStream_F32::release (blockS);
}

149
synth_sine_f32.h
Unescape View File

@ -1,5 +1,5 @@
/*
* AdioSynthWaveformSine_F32
* AudioSynthWaveformSine_F32
*
* Created: Chip Audette (OpenAudio) Feb 2017
* Modeled on: AudioSynthWaveformSine from Teensy Audio Library
@ -9,55 +9,128 @@
* License: MIT License. Use at your own risk.
*
*/
/* Revised 7 Feb 2022 to use a larger 512 point table and direct floating
* point. The level of harmonics depends on the exact frequency, but seems
* to be around -110 dB below the sine wave output. This is more than
* adequate for most applications. For some testing, a pure sine wave,
* limited only by the 24 bit mantissa, is useful. For this, the function
* pureSpectrum(true) will run two stages of biquad filtering putting the
* harmonics below -135 dBc. This filter tracks the frequency() entry, and
* is available above a few hundred Hz, depending on the sample rate. --Bob
*
* Update time is about 9 microsends for 128 update() with T4.x. This goes
* up to 16 microseconds if "pureSpectrum" is used.
*/
#ifndef synth_sine_f32_h_
#define synth_sine_f32_h_
#ifndef synth_sine2_f32_h_
#define synth_sine2_f32_h_
#include "Arduino.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
class AudioSynthWaveformSine_F32 : public AudioStream_F32
{
//GUI: inputs:0, outputs:1 //this line used for automatic generation of GUI node
//GUI: shortName:sine //this line used for automatic generation of GUI node
public:
AudioSynthWaveformSine_F32() : AudioStream_F32(0, NULL), magnitude(16384) { } //uses default AUDIO_SAMPLE_RATE from AudioStream.h
AudioSynthWaveformSine_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL), magnitude(16384) {
setSampleRate_Hz(settings.sample_rate_Hz);
}
void frequency(float freq) {
if (freq < 0.0) freq = 0.0;
else if (freq > sample_rate_Hz/2.f) freq = sample_rate_Hz/2.f;
phase_increment = freq * (4294967296.0 / sample_rate_Hz);
}
void phase(float angle) {
if (angle < 0.0f) angle = 0.0f;
else if (angle > 360.0f) {
angle = angle - 360.0f;
if (angle >= 360.0f) return;
}
phase_accumulator = angle * (4294967296.0f / 360.0f);
}
void amplitude(float n) {
if (n < 0) n = 0;
else if (n > 1.0f) n = 1.0f;
magnitude = n * 65536.0f;
}
void setSampleRate_Hz(const float &fs_Hz) {
phase_increment *= sample_rate_Hz / fs_Hz; //change the phase increment for the new frequency
sample_rate_Hz = fs_Hz;
}
void begin(void) { enabled = true; }
void end(void) { enabled = false; }
virtual void update(void);
AudioSynthWaveformSine_F32() : AudioStream_F32(0, NULL), magnitude(0.5f) {
initSine();
} //uses default AUDIO_SAMPLE_RATE from AudioStream.h
AudioSynthWaveformSine_F32(const AudioSettings_F32 &settings) :
AudioStream_F32(0, NULL), magnitude(0.5f) {
setSampleRate_Hz(settings.sample_rate_Hz);
initSine();
}
void initSine(void) {
for(int ii=0; ii<10; ii++) // Coeff for BiQuad BPF
coeff32[ii] = 0.0;
coeff32[0] = 1.0; // b0 = 1 for pass through
coeff32[5] = 1.0;
// {numStages, pState, pCoeffs};
arm_biquad_cascade_df1_init_f32( &bq_inst, 2, state32, coeff32 );
}
void frequency(float32_t _freq) { // Frequency in Hz
freq = _freq;
if (freq < 0.0f)
freq = 0.0f;
if (freq > sample_rate_Hz/2.0f)
freq = sample_rate_Hz/2.0f;
phaseIncrement = 512.0f * freq / sample_rate_Hz;
// Find coeff for 2 stages of BPF to remove harmoncs
// Always compute these in case pureSpectrum is enabled later.
if(freq > 0.003f*sample_rate_Hz)
{
float32_t q = 20.0f;
float32_t w0 = freq * (2.0f * 3.141592654f / sample_rate_Hz);
float32_t alpha = sin(w0) / (q * 2.0);
float32_t scale = 1.0f / (1.0f + alpha);
/* b0 */ coeff32[0] = alpha * scale;
/* b1 */ coeff32[1] = 0;
/* b2 */ coeff32[2] = (-alpha) * scale;
/* a1 */ coeff32[3] = -(-2.0 * cos(w0)) * scale;
/* a2 */ coeff32[4] = -(1.0 - alpha) * scale;
/* b0 */ coeff32[5] = coeff32[0];
/* b1 */ coeff32[6] = coeff32[1];
/* b2 */ coeff32[7] = coeff32[2];
/* a1 */ coeff32[8] = coeff32[3];
/* a2 */ coeff32[9] = coeff32[4];
arm_biquad_cascade_df1_init_f32( &bq_inst, 2, coeff32, state32 );
}
else
{
for(int ii=0; ii<10; ii++) // Coeff for BiQuad BPF
coeff32[ii] = 0.0;
coeff32[0] = 1.0; // b0 = 1 for pass through
coeff32[5] = 1.0;
arm_biquad_cascade_df1_init_f32( &bq_inst, 2, coeff32, state32 );
enabled = false;
}
}
/* Externally, phase comes in the range (.0, 360.0).
* Internally, the full circle is represented as (0.0, 512.0). This is
* convenient for finding the entry to the sine table.
* Corrected 1-day at phase_r() 24 Feb 22
*/
void phase(float32_t _angle) {
angle = 1.42222222f*_angle; // Change (0,360) to (0, 512)
while (angle < 0.0f) angle += 512.0f;
while (angle > 512.0f) angle -= 512.0;
}
// The amplitude, a, is the peak, as in zero-to-peak. This produces outputs
// ranging from -a to +a.
void amplitude(float32_t a) {
if (a < 0.0f) a = 0.0f;
magnitude = a;
}
void setSampleRate_Hz(const float &fs_Hz) {
phaseIncrement *= sample_rate_Hz / fs_Hz; //change the phase increment for the new frequency
sample_rate_Hz = fs_Hz;
}
void begin(void) { enabled = true; }
void end(void) { enabled = false; }
void pureSpectrum(bool _setPure) { doPureSpectrum = _setPure; }
virtual void update(void);
private:
uint32_t phase_accumulator = 0;
uint32_t phase_increment = 0;
int32_t magnitude = 0;
float sample_rate_Hz = AUDIO_SAMPLE_RATE;
volatile uint8_t enabled = 1;
float32_t freq = 1000.0f;
float32_t angle = 0.0f; // Phase angle
float32_t phaseS = 0.0f;
float32_t phaseIncrement = 0.0f;
float32_t magnitude = 0.0f;
float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE;
bool doPureSpectrum = false; // Adds bandpass filter (not normally needed)
bool enabled = true;
float32_t coeff32[10]; // 2 biquad stages for filtering output
float32_t state32[8];
arm_biquad_casd_df1_inst_f32 bq_inst; // ARM DSP Math library filter instance.
};
#endif

Write Preview
Loading…
Cancel
Save