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OpenAudio_ArduinoLibrary/synth_sine_f32.h

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/*
* AudioSynthWaveformSine_F32
*
* Created: Chip Audette (OpenAudio) Feb 2017
* Modeled on: AudioSynthWaveformSine from Teensy Audio Library
*
* Purpose: Create sine wave of given amplitude and frequency
*
* 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_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(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.
*/
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:
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