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204 lines
7.8 KiB
204 lines
7.8 KiB
/* synth_sin_cos_f32.h
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* AudioSynthSineCosine_F32
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*
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* Status: Checked for function and accuracy. 19 April 2020
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* 10 March 2021 Corrected Interpolation equations. Bob L
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* 23 Feb 2022 Added "pure spectrum" filtering. RSL
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* This adds the function pureSpectrum(bool _setPure); // Default: false
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*
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* Created: Bob Larkin 15 April 2020
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*
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* Based on Chip Audette's OpenAudio sine(), that was
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* Modeled on: AudioSynthWaveformSine from Teensy Audio Library
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*
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* Purpose: Create sine and cosine wave of given amplitude, frequency
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* and phase. Outputs are audio_block_f32_t blocks of float32_t.
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* Routines are from the arm CMSIS library and use a 512 point lookup
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* table with linear interpolation to achieve float accuracy limits.
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*
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* This provides for setting the phase of the sine, setting the difference
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* in phase between the sine and cosine and setting the
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* (-amplitude, amplitude) range. If these are at the default values,
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* called doSimple, the caluclation is faster.
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* For doSimple either true or false, the frequency can be changed
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* at will using the frequency() method.
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*
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* Defaults:
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* Frequency: 1000.0 Hz
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* Phase of Sine: 0.0 radians (0.0 deg)
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* Phase of Cosine: pi/2 radians (90.0 deg) ahead of Sine
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* Amplitude: -1.0 to 1.0
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*
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* Time: T3.6 update() block of 128 with doSimple is 36 microseconds
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* Same using flexible doSimple=false is 49 microseconds
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* T4.0 update() block of 128 with doSimple is 16 microseconds
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* Same using flexible doSimple=false is 24 microseconds
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*
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* Copyright (c) 2020 Bob Larkin
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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#ifndef synth_sin_cos_f32_h_
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#define synth_sin_cos_f32_h_
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#include "AudioStream_F32.h"
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#include "arm_math.h"
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#ifndef M_PI
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#define M_PI 3.14159265358979323846
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#endif
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#ifndef M_PI_2
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#define M_PI_2 1.57079632679489661923
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#endif
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#ifndef M_TWOPI
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#define M_TWOPI (M_PI * 2.0)
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#endif
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#define MF2_PI 6.2831853f
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class AudioSynthSineCosine_F32 : public AudioStream_F32 {
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//GUI: inputs:0, outputs:2 //this line used for automatic generation of GUI node
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//GUI: shortName:SineCosine //this line used for automatic generation of GUI node
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public:
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AudioSynthSineCosine_F32(void) : AudioStream_F32(0, NULL) { } //uses default AUDIO_SAMPLE_RATE from AudioStream.h
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AudioSynthSineCosine_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
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setSampleRate_Hz(settings.sample_rate_Hz);
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setBlockLength(settings.audio_block_samples);
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}
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void frequency(float32_t fr) { // Frequency in Hz
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freq = fr;
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if (freq < 0.0f) freq = 0.0f;
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else if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.0f;
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phaseIncrement = 512.0f * freq / sample_rate_Hz;
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// Find coeff for 2 stages of BPF to remove harmoncs
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// Always compute these in case pureSpectrum is enabled later.
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if(freq > 0.003f*sample_rate_Hz)
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{
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float32_t q = 20.0f;
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float32_t w0 = freq * (2.0f * 3.141592654f / sample_rate_Hz);
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float32_t alpha = sin(w0) / (q * 2.0);
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float32_t scale = 1.0f / (1.0f + alpha);
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/* b0 */ coeff32[0] = alpha * scale;
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/* b1 */ coeff32[1] = 0;
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/* b2 */ coeff32[2] = (-alpha) * scale;
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/* a1 */ coeff32[3] = -(-2.0 * cos(w0)) * scale;
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/* a2 */ coeff32[4] = -(1.0 - alpha) * scale;
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/* b0 */ coeff32[5] = coeff32[0];
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/* b1 */ coeff32[6] = coeff32[1];
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/* b2 */ coeff32[7] = coeff32[2];
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/* a1 */ coeff32[8] = coeff32[3];
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/* a2 */ coeff32[9] = coeff32[4];
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arm_biquad_cascade_df1_init_f32( &bq_instS, 2, coeff32, state32S );
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arm_biquad_cascade_df1_init_f32( &bq_instC, 2, coeff32, state32C );
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}
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else
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{
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for(int ii=0; ii<10; ii++) // Coeff for BiQuad BPF
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coeff32[ii] = 0.0;
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coeff32[0] = 1.0; // b0 = 1 for pass through
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coeff32[5] = 1.0;
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arm_biquad_cascade_df1_init_f32( &bq_instS, 2, coeff32, state32S );
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arm_biquad_cascade_df1_init_f32( &bq_instC, 2, coeff32, state32C );
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}
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}
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/* Externally, phase comes in the range (0,2*M_PI) keeping with C math functions
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* Internally, the full circle is represented as (0.0, 512.0). This is
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* convenient for finding the entry to the sine table.
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*/
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void phase_r(float32_t a) {
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while (a < 0.0f) a += MF2_PI;
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while (a > MF2_PI) a -= MF2_PI;
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phaseS = 512.0f * a / MF2_PI;
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doSimple = false;
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return;
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}
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// phaseS_C_r is the number of radians that the cosine output leads the
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// sine output. The default is M_PI_2 = pi/2 = 1.57079633 radians,
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// corresponding to 90.00 degrees cosine leading sine.
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void phaseS_C_r(float32_t a) {
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while (a < 0.0f) a += MF2_PI;
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while (a > MF2_PI) a -= MF2_PI;
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// Internally a full circle is 512.00 of phase
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phaseS_C = 512.0f * a / MF2_PI;
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doSimple = false;
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return;
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}
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// The amplitude, a, is the peak, as in zero-to-peak. This produces outputs
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// ranging from -a to +a. Both outputs are the same amplitude.
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void amplitude(float32_t a) {
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amplitude_pk = a;
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doSimple = false;
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return;
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}
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// Speed up calculations by setting phaseS_C=90deg, amplitude=1
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// Note, s=true will override any setting of phaseS_C_r or amplitude.
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void simple(bool s) {
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doSimple = s;
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if(doSimple) {
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phaseS_C = 128.0f;
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amplitude_pk = 1.0f;
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}
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return;
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}
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void setSampleRate_Hz(float32_t fs_Hz) {
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// Check freq range
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if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.f;
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// update phase increment for new frequency
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phaseIncrement = 512.0f * freq / fs_Hz;
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}
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void setBlockLength(uint16_t bl) {
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if(bl > 128) bl = 128;
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block_length = bl;
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}
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void pureSpectrum(bool _setPure) { doPureSpectrum = _setPure; }
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virtual void update(void);
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private:
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float32_t freq = 1000.0f;
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float32_t phaseS = 0.0f;
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float32_t phaseS_C = 128.00;
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float32_t amplitude_pk = 1.0f;
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float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE;
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float32_t phaseIncrement = 512.0f * freq /sample_rate_Hz;
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uint16_t block_length = 128;
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// if only freq() is used, the complexities of phase, phaseS_C,
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// and amplitude are not used, speeding up the sin and cos:
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bool doSimple = true;
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bool doPureSpectrum = false; // Adds bandpass filter (not normally needed)
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float32_t coeff32[10]; // 2 biquad stages for filtering output shared S&C
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float32_t state32S[8];
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arm_biquad_casd_df1_inst_f32 bq_instS; // ARM DSP Math library filter instance.
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float32_t state32C[8];
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arm_biquad_casd_df1_inst_f32 bq_instC; // ARM DSP Math library filter instance.
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};
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#endif
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