Remove old version of AudioSynthWaveform. Fix sine wave generation to clean up its specturm and also fix ramping.

synth_waveform.cpp

@@ -29,90 +29,10 @@
 #include "utility/dspinst.h"
 
 
-#ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
 /******************************************************************/
 // PAH - add ramp-up and ramp-down to the onset of the wave
 // the length is specified in samples
-void AudioSynthWaveform::set_ramp_length(uint16_t r_length)
-{
-	if(r_length < 0) {
-		ramp_length = 0;
-		return;
-	}
-	// Don't set the ramp length longer than about 4 milliseconds
-	if(r_length > 44*4) {
-		ramp_length = 44*4;
-		return;
-	}
-	ramp_length = r_length;
-}
-
-void AudioSynthWaveform::update(void)
-{
-	audio_block_t *block;
-	uint32_t i, ph, inc, index, scale;
-	int32_t val1, val2, val3;
-
-	//Serial.println("AudioSynthWaveform::update");
-	if (((magnitude > 0) || ramp_down) && (block = allocate()) != NULL) {
-		ph = phase;
-		inc = phase_increment;
-		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
-			index = ph >> 24;
-			val1 = wavetable[index];
-			val2 = wavetable[index+1];
-			scale = (ph >> 8) & 0xFFFF;
-			val2 *= scale;
-			val1 *= 0xFFFF - scale;
-			val3 = (val1 + val2) >> 16;
-
-
-// The value of ramp_up is always initialized to RAMP_LENGTH and then is
-// decremented each time through here until it reaches zero.
-// The value of ramp_up is used to generate a Q15 fraction which varies
-// from [0 - 1), and multiplies this by the current sample
-			if(ramp_up) {
-				// ramp up to the new magnitude
-				// ramp_mag is the Q15 representation of the fraction
-				// Since ramp_up can't be zero, this cannot generate +1
-				ramp_mag = ((ramp_length-ramp_up)<<15)/ramp_length;
-				ramp_up--;
-				block->data[i] = (val3 * ((ramp_mag * magnitude)>>15)) >> 15;
-
-			} else if(ramp_down) {
-				// ramp down to zero from the last magnitude
-// The value of ramp_down is always initialized to RAMP_LENGTH and then is
-// decremented each time through here until it reaches zero.
-// The value of ramp_down is used to generate a Q15 fraction which varies
-// from (1 - 0], and multiplies this by the current sample
-				// avoid RAMP_LENGTH/RAMP_LENGTH because Q15 format
-				// cannot represent +1
-				ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
-				ramp_down--;
-				block->data[i] = (val3 * ((ramp_mag * last_magnitude)>>15)) >> 15;
-			} else {			
-				block->data[i] = (val3 * magnitude) >> 15;
-			}
-
-			 //Serial.print(block->data[i]);
-			 //Serial.print(", ");
-			 //if ((i % 12) == 11) Serial.println();
-			ph += inc;
-		}
-		 //Serial.println();
-		phase = ph;
-		transmit(block);
-		release(block);
-	} else {
-		// is this numerical overflow ok?
-		phase += phase_increment * AUDIO_BLOCK_SAMPLES;
-	}
-}
-#else
-/******************************************************************/
-// PAH - add ramp-up and ramp-down to the onset of the wave
-// the length is specified in samples
-void AudioSynthWaveform::set_ramp_length(uint16_t r_length)
+void AudioSynthWaveform::set_ramp_length(int16_t r_length)
 {
   if(r_length < 0) {
     ramp_length = 0;
@@ -135,7 +55,8 @@ boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
   if(t_hi < 1)return false;
   if(t_hi >= AUDIO_SAMPLE_RATE_EXACT/2)return false;
   tone_phase = 0;
-  tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
+//  tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
+  tone_incr = (0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
   if(0) {
     Serial.print("AudioSynthWaveform.begin(tone_amp = ");
     Serial.print(t_amp);
@@ -150,7 +71,9 @@ boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
   return(true);
 }
 
-
+// PAH - 140313 fixed the calculation of the tone so that its spectrum
+//              is much improved
+// PAH - 140313 fixed a problem with ramping
 void AudioSynthWaveform::update(void)
 {
   audio_block_t *block;
@@ -182,31 +105,35 @@ void AudioSynthWaveform::update(void)
           ramp_up--;
           // adjust tone_phase to Q15 format and then adjust the result
           // of the multiplication
-          *bp = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15);
-          *bp++ = (*bp * ramp_mag)>>15;
+      	// calculate the sample
+          tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
+          *bp++ = (tmp_amp * ramp_mag)>>15;
         } 
         else if(ramp_down) {
           // ramp down to zero from the last magnitude
           // The value of ramp_down is always initialized to RAMP_LENGTH and then is
           // decremented each time through here until it reaches zero.
           // The value of ramp_down is used to generate a Q15 fraction which varies
-          // from (1 - 0], and multiplies this by the current sample
+          // from [0 - 1), and multiplies this by the current sample
           // avoid RAMP_LENGTH/RAMP_LENGTH because Q15 format
           // cannot represent +1
           ramp_mag = ((ramp_down - 1)<<15)/ramp_length;
           ramp_down--;
           // adjust tone_phase to Q15 format and then adjust the result
           // of the multiplication
-          *bp = (short)((arm_sin_q15(tone_phase>>17) * last_tone_amp) >> 15);
-          *bp++ = (*bp * ramp_mag)>>15;
+          tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * last_tone_amp) >> 17);
+          *bp++ = (tmp_amp * ramp_mag)>>15;
         } else {
           // adjust tone_phase to Q15 format and then adjust the result
           // of the multiplication
-          *bp++ = (short)((arm_sin_q15(tone_phase>>17) * tone_amp) >> 15);
+          tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
+          *bp++ = tmp_amp;
         } 
         
         // 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;
       
@@ -254,88 +181,3 @@ void AudioSynthWaveform::update(void)
     release(block);
   }
 }
-
-
-#endif
-
-
-
-
-
-
-
-
-#if 0
-void AudioSineWaveMod::frequency(float f)
-{
-	if (f > AUDIO_SAMPLE_RATE_EXACT / 2 || f < 0.0) return;
-	phase_increment = (f / AUDIO_SAMPLE_RATE_EXACT) * 4294967296.0f;
-}
-
-void AudioSineWaveMod::update(void)
-{
-	audio_block_t *block, *modinput;
-	uint32_t i, ph, inc, index, scale;
-	int32_t val1, val2;
-
-	//Serial.println("AudioSineWave::update");
-	modinput = receiveReadOnly();
-	ph = phase;
-	inc = phase_increment;
-	block = allocate();
-	if (!block) {
-		// unable to allocate memory, so we'll send nothing
-		if (modinput) {
-			// but if we got modulation data, update the phase
-			for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
-				ph += inc + modinput->data[i] * modulation_factor;
-			}
-			release(modinput);
-		} else {
-			ph += phase_increment * AUDIO_BLOCK_SAMPLES;
-		}
-		phase = ph;
-		return;
-	}
-	if (modinput) {
-		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
-			index = ph >> 24;
-			val1 = sine_table[index];
-			val2 = sine_table[index+1];
-			scale = (ph >> 8) & 0xFFFF;
-			val2 *= scale;
-			val1 *= 0xFFFF - scale;
-			block->data[i] = (val1 + val2) >> 16;
-			 //Serial.print(block->data[i]);
-			 //Serial.print(", ");
-			 //if ((i % 12) == 11) Serial.println();
-			ph += inc + modinput->data[i] * modulation_factor;
-		}
-		release(modinput);
-	} else {
-		ph = phase;
-		inc = phase_increment;
-		for (i=0; i < AUDIO_BLOCK_SAMPLES; i++) {
-			index = ph >> 24;
-			val1 = sine_table[index];
-			val2 = sine_table[index+1];
-			scale = (ph >> 8) & 0xFFFF;
-			val2 *= scale;
-			val1 *= 0xFFFF - scale;
-			block->data[i] = (val1 + val2) >> 16;
-			 //Serial.print(block->data[i]);
-			 //Serial.print(", ");
-			 //if ((i % 12) == 11) Serial.println();
-			ph += inc;
-		}
-	}
-	phase = ph;
-	transmit(block);
-	release(block);
-}
-#endif
-
-
-
-
-

synth_waveform.h

@@ -30,58 +30,6 @@
 #include "AudioStream.h"
 #include "arm_math.h"
 
-// waveforms.c
-extern "C" {
-extern const int16_t AudioWaveformSine[257];
-extern const int16_t AudioWaveformTriangle[257];
-extern const int16_t AudioWaveformSquare[257];
-extern const int16_t AudioWaveformSawtooth[257];
-}
-
-#ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
-class AudioSynthWaveform : public AudioStream
-{
-public:
-	AudioSynthWaveform(const int16_t *waveform)
-	  : AudioStream(0, NULL), wavetable(waveform), magnitude(0), phase(0)
-					, ramp_down(0), ramp_up(0), ramp_mag(0), ramp_length(0)
-	  				 { }
-	void frequency(float freq) {
-		if (freq > AUDIO_SAMPLE_RATE_EXACT / 2 || freq < 0.0) return;
-		phase_increment = (freq / AUDIO_SAMPLE_RATE_EXACT) * 4294967296.0f;
-	}
-	void amplitude(float n) {        // 0 to 1.0
-		if (n < 0) n = 0;
-		else if (n > 1.0) n = 1.0;
-// Ramp code
-		if(magnitude && (n == 0)) {
-			ramp_down = ramp_length;
-			ramp_up = 0;
-			last_magnitude = magnitude;
-		}
-		else if((magnitude == 0) && n) {
-			ramp_up = ramp_length;
-			ramp_down = 0;
-		}
-// set new magnitude
-		magnitude = n * 32767.0;
-	}
-	virtual void update(void);
-	void set_ramp_length(uint16_t r_length);
-	
-private:
-	const int16_t *wavetable;
-	uint16_t magnitude;
-	uint16_t last_magnitude;
-	uint32_t phase;
-	uint32_t phase_increment;
-	uint32_t ramp_down;
-	uint32_t ramp_up;
-	uint32_t ramp_mag;
-	uint16_t ramp_length;
-};
-
-#else
 
 #define AUDIO_SAMPLE_RATE_ROUNDED (44118)
 
@@ -106,7 +54,8 @@ public:
   // change between two frequencies.
   void frequency(int t_hi)
   {
-    tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;  
+//    tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
+    tone_incr = (0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
   }
   // If ramp_length is non-zero this will set up
   // either a rmap up or a ramp down when a wave
@@ -138,7 +87,7 @@ public:
   }
   boolean begin(float t_amp,int t_hi,short t_type);
   virtual void update(void);
-  void set_ramp_length(uint16_t r_length);
+  void set_ramp_length(int16_t r_length);
 
 private:
   short    tone_amp;
@@ -152,27 +101,4 @@ private:
   uint32_t ramp_up;
   uint16_t ramp_length;
 };
-
-#endif
-
-
-
-#if 0
-class AudioSineWaveMod : public AudioStream
-{
-public:
-	AudioSineWaveMod() : AudioStream(1, inputQueueArray) {}
-	void frequency(float freq);
-	//void amplitude(q15 n);
-	virtual void update(void);
-private:
-	uint32_t phase;
-	uint32_t phase_increment;
-	uint32_t modulation_factor;
-	audio_block_t *inputQueueArray[1];
-};
-#endif
-
-
-
 #endif