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FP-Audio-lib/synth_dc_f32.h

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/* Audio Library for Teensy 3.X
* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* 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, development funding 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_dc_f32_h_
#define synth_dc_f32_h_
#include "Arduino.h"
#include "AudioStream_f32.h"
#include "utility/dspinst.h"
#include "arm_math.h"
// compute (a - b) / c
// handling 32 bit interger overflow at every step
// without resorting to slow 64 bit math
class AudioSynthWaveformDc_F32 : public AudioStream_F32
{
public:
AudioSynthWaveformDc_F32() : AudioStream_F32(0, NULL), state(0), magnitude(0) {}
// immediately jump to the new DC level
void amplitude(float n) {
if (n > 1.0) n = 1.0;
else if (n < -1.0) n = -1.0;
// int i;
// int32_t m = (int32_t)(n * 0x7fff);
__disable_irq();
magnitude = n;
state = 0;
// incblock = allocate_f32();
// for(i=0;i<AUDIO_BLOCK_SAMPLES;i++)
// incblock->data[i]=i+1;
__enable_irq();
}
// slowly transition to the new DC level
void amplitude(float n, float milliseconds) {
if (milliseconds <= 0.0) {
amplitude(n);
return;
}
if (n > 1.0) n = 1.0;
else if (n < -1.0) n = -1.0;
float c = (milliseconds*(AUDIO_SAMPLE_RATE_EXACT/1000.0));
if (c == 0) {
amplitude(n);
return;
}
// int32_t t = (int32_t)(n * 0x7fff);
__disable_irq();
target = n;
if (target == magnitude) {
state = 0;
__enable_irq();
return;
}
increment = (target-magnitude)/ c;
if (increment == 0) {
increment = (target > magnitude) ? 1 : -1;
}
state = 1;
__enable_irq();
}
float read(void) {
return magnitude;
}
virtual void update(void);
private:
uint8_t state; // 0=steady output, 1=transitioning
float magnitude; // current output
float target; // designed output (while transitiong)
float increment; // adjustment per sample (while transitiong)
// audio_block_f32_t *incblock;
};
#endif