/* synth_sin_cos_f32.h
* AudioSynthSineCosine_F32
*
* Status: Checked for function and accuracy. 19 April 2020
* 10 March 2021 Corrected Interpolation equations. Bob L
* 23 Feb 2022 Added "pure spectrum" filtering. RSL
* This adds the function pureSpectrum(bool _setPure); // Default: false
*
* Created: Bob Larkin 15 April 2020
*
* Based on Chip Audette's OpenAudio sine(), that was
* Modeled on: AudioSynthWaveformSine from Teensy Audio Library
*
* Purpose: Create sine and cosine wave of given amplitude, frequency
* and phase. Outputs are audio_block_f32_t blocks of float32_t.
* Routines are from the arm CMSIS library and use a 512 point lookup
* table with linear interpolation to achieve float accuracy limits.
*
* This provides for setting the phase of the sine, setting the difference
* in phase between the sine and cosine and setting the
* (-amplitude, amplitude) range. If these are at the default values,
* called doSimple, the caluclation is faster.
* For doSimple either true or false, the frequency can be changed
* at will using the frequency() method.
*
* Defaults:
* Frequency: 1000.0 Hz
* Phase of Sine: 0.0 radians (0.0 deg)
* Phase of Cosine: pi/2 radians (90.0 deg) ahead of Sine
* Amplitude: -1.0 to 1.0
*
* Time: T3.6 update() block of 128 with doSimple is 36 microseconds
* Same using flexible doSimple=false is 49 microseconds
* T4.0 update() block of 128 with doSimple is 16 microseconds
* Same using flexible doSimple=false is 24 microseconds
*
* Copyright (c) 2020 Bob Larkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
#ifndef synth_sin_cos_f32_h_
#define synth_sin_cos_f32_h_
#include "AudioStream_F32.h"
#include "arm_math.h"
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
#ifndef M_PI_2
#define M_PI_2 1.57079632679489661923
#endif
#ifndef M_TWOPI
#define M_TWOPI (M_PI * 2.0)
#endif
#define MF2_PI 6.2831853f
class AudioSynthSineCosine_F32 : public AudioStream_F32 {
//GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI node
//GUI: shortName:SineCosine //this line used for automatic generation of GUI node
public:
AudioSynthSineCosine_F32(void) : AudioStream_F32(0, NULL) { } //uses default AUDIO_SAMPLE_RATE from AudioStream.h
AudioSynthSineCosine_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
setSampleRate_Hz(settings.sample_rate_Hz);
setBlockLength(settings.audio_block_samples);
}
void frequency(float32_t fr) { // Frequency in Hz
freq = fr;
if (freq < 0.0f) freq = 0.0f;
else 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_instS, 2, coeff32, state32S );
arm_biquad_cascade_df1_init_f32( &bq_instC, 2, coeff32, state32C );
}
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_instS, 2, coeff32, state32S );
arm_biquad_cascade_df1_init_f32( &bq_instC, 2, coeff32, state32C );
}
}
/* Externally, phase comes in the range (0,2*M_PI) keeping with C math functions
* Internally, the full circle is represented as (0.0, 512.0). This is
* convenient for finding the entry to the sine table.
*/
void phase_r(float32_t a) {
while (a < 0.0f) a += MF2_PI;
while (a > MF2_PI) a -= MF2_PI;
phaseS = 512.0f * a / MF2_PI;
doSimple = false;
return;
}
// phaseS_C_r is the number of radians that the cosine output leads the
// sine output. The default is M_PI_2 = pi/2 = 1.57079633 radians,
// corresponding to 90.00 degrees cosine leading sine.
void phaseS_C_r(float32_t a) {
while (a < 0.0f) a += MF2_PI;
while (a > MF2_PI) a -= MF2_PI;
// Internally a full circle is 512.00 of phase
phaseS_C = 512.0f * a / MF2_PI;
doSimple = false;
return;
}
// The amplitude, a, is the peak, as in zero-to-peak. This produces outputs
// ranging from -a to +a. Both outputs are the same amplitude.
void amplitude(float32_t a) {
amplitude_pk = a;
doSimple = false;
return;
}
// Speed up calculations by setting phaseS_C=90deg, amplitude=1
// Note, s=true will override any setting of phaseS_C_r or amplitude.
void simple(bool s) {
doSimple = s;
if(doSimple) {
phaseS_C = 128.0f;
amplitude_pk = 1.0f;
}
return;
}
void setSampleRate_Hz(float32_t fs_Hz) {
// Check freq range
if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.f;
// update phase increment for new frequency
phaseIncrement = 512.0f * freq / fs_Hz;
}
void setBlockLength(uint16_t bl) {
if(bl > 128) bl = 128;
block_length = bl;
}
void pureSpectrum(bool _setPure) { doPureSpectrum = _setPure; }
virtual void update(void);
private:
float32_t freq = 1000.0f;
float32_t phaseS = 0.0f;
float32_t phaseS_C = 128.00;
float32_t amplitude_pk = 1.0f;
float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE;
float32_t phaseIncrement = 512.0f * freq /sample_rate_Hz;
uint16_t block_length = 128;
// if only freq() is used, the complexities of phase, phaseS_C,
// and amplitude are not used, speeding up the sin and cos:
bool doSimple = true;
bool doPureSpectrum = false; // Adds bandpass filter (not normally needed)
float32_t coeff32[10]; // 2 biquad stages for filtering output shared S&C
float32_t state32S[8];
arm_biquad_casd_df1_inst_f32 bq_instS; // ARM DSP Math library filter instance.
float32_t state32C[8];
arm_biquad_casd_df1_inst_f32 bq_instC; // ARM DSP Math library filter instance.
};
#endif