Add arbitrary waveform and offset to AudioSynthWaveform

keywords.txt

 updateCoefs	KEYWORD2
 frequency	KEYWORD2
 amplitude	KEYWORD2
+offset	KEYWORD2
 resonance	KEYWORD2
 octaveControl	KEYWORD2
 modify	KEYWORD2
 FILTER_LOSHELF	LITERAL1
 FILTER_HISHELF	LITERAL1
 
+WAVEFORM_SINE	LITERAL1
+WAVEFORM_SAWTOOTH	LITERAL1
+WAVEFORM_SQUARE	LITERAL1
+WAVEFORM_TRIANGLE	LITERAL1
+WAVEFORM_ARBITRARY	LITERAL1
 

synth_sine.cpp

  * THE SOFTWARE.
  */
 
-#include "Audio.h"
-#include "arm_math.h"
+#include "synth_sine.h"
 #include "utility/dspinst.h"
 
 // data_waveforms.c

synth_sine.h

 #ifndef synth_sine_h_
 #define synth_sine_h_
 
+#include "AudioStream.h"
+#include "arm_math.h"
+
 // TODO: investigate making a high resolution sine wave
 // using Taylor series expansion.
 // http://www.musicdsp.org/showone.php?id=13

synth_waveform.cpp

 // PAH 140314 - change t_hi from int to float
 
 
-boolean AudioSynthWaveform::begin(float t_amp,float t_hi,short type)
-{
-  tone_type = type;
-  amplitude(t_amp);
-  tone_freq = t_hi > 0.0;
-  if(t_hi <= 0.0)return false;
-  if(t_hi >= AUDIO_SAMPLE_RATE_EXACT/2)return false;
-  tone_phase = 0;
-  frequency(t_hi);
-  if(0) {
-    Serial.print("AudioSynthWaveform.begin(tone_amp = ");
-    Serial.print(t_amp);
-    Serial.print(", tone_hi = ");
-    Serial.print(t_hi);
-    Serial.print(", tone_incr = ");
-    Serial.print(tone_incr,HEX);
-    //  Serial.print(", tone_hi = ");
-    //  Serial.print(t_hi);
-    Serial.println(")");
-  }
-  return(true);
-}
-
 void AudioSynthWaveform::update(void)
 {
   audio_block_t *block;
-  short *bp;
+  short *bp, *end;
   int32_t val1, val2, val3;
   uint32_t index, scale;
   
   uint32_t mag;
   short tmp_amp;
   
-  if(tone_freq == 0)return;
+  if(tone_amp == 0) return;
   block = allocate();
-  if(block) {
+  if (block) {
     bp = block->data;
     switch(tone_type) {
-    case TONE_TYPE_SINE:
+    case WAVEFORM_SINE:
       for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
       	// Calculate interpolated sin
 		index = tone_phase >> 23;
         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 TONE_TYPE_SQUARE:
+    case WAVEFORM_SQUARE:
       for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
         if(tone_phase & 0x40000000)*bp++ = -tone_amp;
         else *bp++ = tone_amp;
       }
       break;
       
-    case TONE_TYPE_SAWTOOTH:
+    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
       }
       break;
 
-    case TONE_TYPE_TRIANGLE:
+    case WAVEFORM_TRIANGLE:
       for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
         if(tone_phase & 0x80000000) {
           // negative half-cycle
       }
       break;
     }
-    // send the samples to the left channel
+    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

 
 #define DELAY_PASSTHRU -1
 
+#define WAVEFORM_SINE      0
+#define WAVEFORM_SAWTOOTH  1
+#define WAVEFORM_SQUARE    2
+#define WAVEFORM_TRIANGLE  3
+#define WAVEFORM_ARBITRARY 4
+
+// todo: remove these...
 #define TONE_TYPE_SINE     0
 #define TONE_TYPE_SAWTOOTH 1
 #define TONE_TYPE_SQUARE   2
 public:
   AudioSynthWaveform(void) : 
   AudioStream(0,NULL), 
-  tone_freq(0), tone_phase(0), tone_incr(0), tone_type(0)
+  tone_phase(0), tone_incr(0), tone_type(0),
+  tone_offset(0), arbdata(NULL)
   { 
   }
   
-  void frequency(float t_hi)
-  {
-    if (t_hi > AUDIO_SAMPLE_RATE_EXACT / 2 || t_hi < 0.0) return;
-    tone_incr = ((0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT) + 0.5;
+  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 amplitude(float n) {        // 0 to 1.0
     if (n < 0) n = 0;
     else if (n > 1.0) n = 1.0;
-    // Ramp code
-    if(tone_amp && (n == 0)) {
-      last_tone_amp = tone_amp;
-    }
-    else if((tone_amp == 0) && n) {
+    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;
   }
-  
-  boolean begin(float t_amp,float t_hi,short t_type);
+  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 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);
+  }
   virtual void update(void);
   
 private:
   short    tone_amp;
-  short    last_tone_amp;
   short    tone_freq;
   uint32_t tone_phase;
   // 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;
 };