Reformat AudioSynthWaveform (no actual code change)

synth_waveform.cpp

@@ -1,5 +1,5 @@
 /* Audio Library for Teensy 3.X
- * Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
+ * Copyright (c) 2018, 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
@@ -39,136 +39,138 @@
 
 void AudioSynthWaveform::update(void)
 {
-  audio_block_t *block;
-  short *bp, *end;
-  int32_t val1, val2, val3;
-  uint32_t index, scale;
-  
-  // temporaries for TRIANGLE
-  uint32_t mag;
-  short tmp_amp;
-  
-  if(tone_amp == 0) return;
-  block = allocate();
-  if (block) {
-    bp = block->data;
-    switch(tone_type) {
-    case WAVEFORM_SINE:
-      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-      	// Calculate interpolated sin
-		index = tone_phase >> 23;
-		val1 = AudioWaveformSine[index];
-		val2 = AudioWaveformSine[index+1];
-		scale = (tone_phase >> 7) & 0xFFFF;
-		val2 *= scale;
-		val1 *= 0xFFFF - scale;
-		val3 = (val1 + val2) >> 16;
-		*bp++ = (short)((val3 * tone_amp) >> 15);
-        
-        // phase and incr are both unsigned 32-bit fractions
-        tone_phase += tone_incr;
-        // If tone_phase has overflowed, truncate the top bit 
-        if(tone_phase & 0x80000000)tone_phase &= 0x7fffffff;
-      }
-      break;
-
-    case WAVEFORM_ARBITRARY:
-      if (!arbdata) {
-		release(block);
+	audio_block_t *block;
+	short *bp, *end;
+	int32_t val1, val2, val3;
+	uint32_t index, scale;
+	uint32_t mag;
+	short tmp_amp;
+
+	if (tone_amp == 0) return;
+	block = allocate();
+	if (!block) {
 		return;
-      }
-      // len = 256
-      for (int i = 0; i < AUDIO_BLOCK_SAMPLES;i++) {
-		index = tone_phase >> 23;
-		val1 = *(arbdata + index);
-		val2 = *(arbdata + ((index + 1) & 255));
-		scale = (tone_phase >> 7) & 0xFFFF;
-		val2 *= scale;
-		val1 *= 0xFFFF - scale;
-		val3 = (val1 + val2) >> 16;
-		*bp++ = (short)((val3 * tone_amp) >> 15);
-		tone_phase += tone_incr;
-		tone_phase &= 0x7fffffff;
-      }
-      break;
-      
-    case WAVEFORM_SQUARE:
-      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        if(tone_phase & 0x40000000)*bp++ = -tone_amp;
-        else *bp++ = tone_amp;
-        // phase and incr are both unsigned 32-bit fractions
-        tone_phase += tone_incr;
-      }
-      break;
-      
-    case WAVEFORM_SAWTOOTH:
-      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        *bp++ = ((short)(tone_phase>>15)*tone_amp) >> 15;
-        // phase and incr are both unsigned 32-bit fractions
-        tone_phase += tone_incr;    
-      }
-      break;
-
-    case WAVEFORM_SAWTOOTH_REVERSE:
-      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        *bp++ = ((short)(tone_phase>>15)*tone_amp) >> 15;
-         // phase and incr are both unsigned 32-bit fractions
-         tone_phase -= tone_incr;
-      }
-      break;
-
-    case WAVEFORM_TRIANGLE:
-      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        if(tone_phase & 0x80000000) {
-          // negative half-cycle
-          tmp_amp = -tone_amp;
-        } 
-        else {
-          // positive half-cycle
-          tmp_amp = tone_amp;
-        }
-        mag = tone_phase << 2;
-        // Determine which quadrant
-        if(tone_phase & 0x40000000) {
-          // negate the magnitude
-          mag = ~mag + 1;
-        }
-        *bp++ = ((short)(mag>>17)*tmp_amp) >> 15;
-        tone_phase += 2*tone_incr;
-      }
-      break;
-      
-    case WAVEFORM_PULSE:
-      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        if(tone_phase < tone_width)*bp++ = -tone_amp;
-        else *bp++ = tone_amp;
-        tone_phase += tone_incr;
-      }
-      break;
-      
-    case WAVEFORM_SAMPLE_HOLD:
-      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        if(tone_phase < tone_incr) {
-          sample = random(-tone_amp, tone_amp);
-        }
-        *bp++ = sample;
-        tone_phase += tone_incr;
-      }
-      break;
-    }
-    if (tone_offset) {
+	}
 	bp = block->data;
-	end = bp + AUDIO_BLOCK_SAMPLES;
-	do {
-		val1 = *bp;
-		*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
-	} while (bp < end);
-    }
-    transmit(block,0);
-    release(block);
-  }
-}
+	switch(tone_type) {
+	case WAVEFORM_SINE:
+		for(int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			// Calculate interpolated sin
+			index = tone_phase >> 23;
+			val1 = AudioWaveformSine[index];
+			val2 = AudioWaveformSine[index+1];
+			scale = (tone_phase >> 7) & 0xFFFF;
+			val2 *= scale;
+			val1 *= 0xFFFF - scale;
+			val3 = (val1 + val2) >> 16;
+			*bp++ = (short)((val3 * tone_amp) >> 15);
+
+			// phase and incr are both unsigned 32-bit fractions
+			tone_phase += tone_incr;
+			// If tone_phase has overflowed, truncate the top bit
+			if (tone_phase & 0x80000000) tone_phase &= 0x7fffffff;
+		}
+		break;
+
+	case WAVEFORM_ARBITRARY:
+		if (!arbdata) {
+			release(block);
+			return;
+		}
+		// len = 256
+		for (int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			index = tone_phase >> 23;
+			val1 = *(arbdata + index);
+			val2 = *(arbdata + ((index + 1) & 255));
+			scale = (tone_phase >> 7) & 0xFFFF;
+			val2 *= scale;
+			val1 *= 0xFFFF - scale;
+			val3 = (val1 + val2) >> 16;
+			*bp++ = (short)((val3 * tone_amp) >> 15);
+			tone_phase += tone_incr;
+			tone_phase &= 0x7fffffff;
+		}
+		break;
+
+	case WAVEFORM_SQUARE:
+		for(int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			if (tone_phase & 0x40000000) {
+				*bp++ = -tone_amp;
+			} else {
+				*bp++ = tone_amp;
+			}
+			// phase and incr are both unsigned 32-bit fractions
+			tone_phase += tone_incr;
+		}
+		break;
 
+	case WAVEFORM_SAWTOOTH:
+		for(int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			*bp++ = ((short)(tone_phase>>15) * tone_amp) >> 15;
+			// phase and incr are both unsigned 32-bit fractions
+			tone_phase += tone_incr;
+		}
+		break;
 
+	case WAVEFORM_SAWTOOTH_REVERSE:
+		for(int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			*bp++ = ((short)(tone_phase>>15) * tone_amp) >> 15;
+			// phase and incr are both unsigned 32-bit fractions
+			tone_phase -= tone_incr;
+		}
+		break;
+
+	case WAVEFORM_TRIANGLE:
+		for(int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			if (tone_phase & 0x80000000) {
+				// negative half-cycle
+				tmp_amp = -tone_amp;
+			} else {
+				// positive half-cycle
+				tmp_amp = tone_amp;
+			}
+			mag = tone_phase << 2;
+			// Determine which quadrant
+			if (tone_phase & 0x40000000) {
+				// negate the magnitude
+				mag = ~mag + 1;
+			}
+			*bp++ = ((short)(mag>>17) * tmp_amp) >> 15;
+			tone_phase += tone_incr * 2;
+		}
+		break;
+
+	case WAVEFORM_PULSE:
+		for(int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			if (tone_phase < tone_width) {
+				*bp++ = -tone_amp;
+			} else {
+				*bp++ = tone_amp;
+			}
+			tone_phase += tone_incr;
+		}
+		break;
+
+	case WAVEFORM_SAMPLE_HOLD:
+		for(int i = 0; i < AUDIO_BLOCK_SAMPLES; i++) {
+			if (tone_phase < tone_incr) {
+				sample = random(-tone_amp, tone_amp);
+			}
+			*bp++ = sample;
+			tone_phase += tone_incr;
+		}
+		break;
+	}
+	if (tone_offset) {
+		bp = block->data;
+		end = bp + AUDIO_BLOCK_SAMPLES;
+		do {
+			val1 = *bp;
+			*bp++ = signed_saturate_rshift(val1 + tone_offset, 16, 0);
+		} while (bp < end);
+	}
+	transmit(block, 0);
+	release(block);
+}
 
 

synth_waveform.h

@@ -27,7 +27,7 @@
 #ifndef synth_waveform_h_
 #define synth_waveform_h_
 
-#include "Arduino.h"
+#include <Arduino.h>
 #include "AudioStream.h"
 #include "arm_math.h"
 
@@ -36,9 +36,6 @@ extern "C" {
 extern const int16_t AudioWaveformSine[257];
 }
 
-#define AUDIO_SAMPLE_RATE_ROUNDED (44118)
-
-#define DELAY_PASSTHRU -1
 
 #define WAVEFORM_SINE      0
 #define WAVEFORM_SAWTOOTH  1
@@ -49,84 +46,89 @@ extern const int16_t AudioWaveformSine[257];
 #define WAVEFORM_SAWTOOTH_REVERSE 6
 #define WAVEFORM_SAMPLE_HOLD 7
 
-// todo: remove these...
-#define TONE_TYPE_SINE     0
-#define TONE_TYPE_SAWTOOTH 1
-#define TONE_TYPE_SQUARE   2
-#define TONE_TYPE_TRIANGLE 3
 
-class AudioSynthWaveform : 
-public AudioStream
+class AudioSynthWaveform : public AudioStream
 {
 public:
-  AudioSynthWaveform(void) : 
-  AudioStream(0,NULL), tone_amp(0), tone_freq(0),
-  tone_phase(0), tone_width(0.25), tone_incr(0), tone_type(0),
-  tone_offset(0), arbdata(NULL)
-  { 
-  }
-  
-  void frequency(float t_freq) {
-    if (t_freq < 0.0) t_freq = 0.0;
-    else if (t_freq > AUDIO_SAMPLE_RATE_EXACT / 2) t_freq = AUDIO_SAMPLE_RATE_EXACT / 2;
-    tone_incr = (t_freq * (0x80000000LL/AUDIO_SAMPLE_RATE_EXACT)) + 0.5;
-  }
-  void phase(float angle) {
-    if (angle < 0.0) angle = 0.0;
-    else if (angle > 360.0) {
-      angle = angle - 360.0;
-      if (angle >= 360.0) return;
-    }
-    tone_phase = angle * (2147483648.0 / 360.0);
-  }
-  void amplitude(float n) {        // 0 to 1.0
-    if (n < 0) n = 0;
-    else if (n > 1.0) n = 1.0;
-    if ((tone_amp == 0) && n) {
-      // reset the phase when the amplitude was zero
-      // and has now been increased.
-      tone_phase = 0;
-    }
-    // set new magnitude
-    tone_amp = n * 32767.0;
-  }
-  void offset(float n) {
-    if (n < -1.0) n = -1.0;
-    else if (n > 1.0) n = 1.0;
-    tone_offset = n * 32767.0;
-  }
-  void pulseWidth(float n) {          // 0.0 to 1.0
-    if (n < 0) n = 0;
-    else if (n > 1.0) n = 1.0;
-    tone_width = n * 0x7fffffffLL; 
-    // pulse width is stored as the equivalent phase
-  }
-  void begin(short t_type) {
-	tone_phase = 0;
-	tone_type = t_type;
-  }
-  void begin(float t_amp, float t_freq, short t_type) {
-	amplitude(t_amp);
-	frequency(t_freq);
-	begin(t_type);
-  }
-  void arbitraryWaveform(const int16_t *data, float maxFreq) {
-	arbdata = data;
-  }
-  virtual void update(void);
-  
+	AudioSynthWaveform(void) : AudioStream(0,NULL), tone_amp(0), tone_freq(0),
+		tone_phase(0), tone_width(0.25), tone_incr(0), tone_type(0),
+		tone_offset(0), arbdata(NULL) {
+	}
+
+	void frequency(float t_freq) {
+		if (t_freq < 0.0) {
+			t_freq = 0.0;
+		} else if (t_freq > AUDIO_SAMPLE_RATE_EXACT / 2) {
+			t_freq = AUDIO_SAMPLE_RATE_EXACT / 2;
+		}
+		tone_incr = (t_freq * (0x80000000LL/AUDIO_SAMPLE_RATE_EXACT)) + 0.5;
+	}
+	void phase(float angle) {
+		if (angle < 0.0) {
+			angle = 0.0;
+		} else if (angle > 360.0) {
+			angle = angle - 360.0;
+			if (angle >= 360.0) return;
+		}
+		tone_phase = angle * (2147483648.0 / 360.0);
+	}
+	void amplitude(float n) {	// 0 to 1.0
+		if (n < 0) {
+			n = 0;
+		} else if (n > 1.0) {
+			n = 1.0;
+		}
+		if ((tone_amp == 0) && n) {
+			// reset the phase when the amplitude was zero
+			// and has now been increased.
+			tone_phase = 0;
+		}
+		// set new magnitude
+		tone_amp = n * 32767.0;
+	}
+	void offset(float n) {
+		if (n < -1.0) {
+			n = -1.0;
+		} else if (n > 1.0) {
+			n = 1.0;
+		}
+		tone_offset = n * 32767.0;
+	}
+	void pulseWidth(float n) {	// 0.0 to 1.0
+		if (n < 0) {
+			n = 0;
+		} else if (n > 1.0) {
+			n = 1.0;
+		}
+		tone_width = n * 0x7fffffffLL;
+		// pulse width is stored as the equivalent phase
+	}
+	void begin(short t_type) {
+		tone_phase = 0;
+		tone_type = t_type;
+	}
+	void begin(float t_amp, float t_freq, short t_type) {
+		amplitude(t_amp);
+		frequency(t_freq);
+		begin(t_type);
+	}
+	void arbitraryWaveform(const int16_t *data, float maxFreq) {
+		arbdata = data;
+	}
+	virtual void update(void);
+
 private:
-  short    tone_amp;
-  short    tone_freq;
-  uint32_t tone_phase;
-  uint32_t tone_width;
-  // sample for SAMPLE_HOLD
-  short sample;
-  // volatile prevents the compiler optimizing out the frequency function
-  volatile uint32_t tone_incr;
-  short    tone_type;
-  int16_t  tone_offset;
-  const int16_t *arbdata;
+	short    tone_amp;
+	short    tone_freq;
+	uint32_t tone_phase;
+	uint32_t tone_width;
+	// sample for SAMPLE_HOLD
+	short sample;
+	// volatile prevents the compiler optimizing out the frequency function
+	volatile uint32_t tone_incr;
+	short    tone_type;
+	int16_t  tone_offset;
+	const int16_t *arbdata;
 };