Never allow freq modulation to attempt higher than Nyquist

synth_waveform.cpp

@@ -30,6 +30,10 @@
 #include "utility/dspinst.h"
 
 
+// uncomment for more accurate but more computationally expensive frequency modulation
+//#define IMPROVE_EXPONENTIAL_ACCURACY
+
+
 void AudioSynthWaveform::update(void)
 {
 	audio_block_t *block;
@@ -213,6 +217,7 @@ void AudioSynthWaveformModulated::update(void)
 			n = multiply_accumulate_32x32_rshift32_rounded(n, sq, 1934101615);
 			n = n + (multiply_32x32_rshift32_rounded(sq,
 				multiply_32x32_rshift32_rounded(x, 1358044250)) << 1);
+			n = n << 1;
 			#else
 			// exp2 algorithm by Laurent de Soras
 			// http://www.musicdsp.org/showone.php?id=106
@@ -222,8 +227,13 @@ void AudioSynthWaveformModulated::update(void)
 			n = n + 715827882;
 			#endif
 			uint32_t scale = n >> (14 - ipart);
-			uint32_t phinc = ((uint64_t)inc * scale) >> 16; // TODO: saturate 31 bits??
-			ph += phinc;
+			uint64_t phstep = (uint64_t)inc * scale;
+			uint32_t phstep_msw = phstep >> 32;
+			if (phstep_msw < 0x7FFE) {
+				ph += phstep >> 16;
+			} else {
+				ph += 0x7FFE0000;
+			}
 			phasedata[i] = ph;
 		}
 		release(moddata);

synth_waveform.h

@@ -64,6 +64,7 @@ public:
 			freq = AUDIO_SAMPLE_RATE_EXACT / 2;
 		}
 		phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
+		if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
 	}
 	void phase(float angle) {
 		if (angle < 0.0) {
@@ -131,7 +132,7 @@ class AudioSynthWaveformModulated : public AudioStream
 public:
 	AudioSynthWaveformModulated(void) : AudioStream(2, inputQueueArray),
 		phase_accumulator(0), phase_increment(0), modulation_factor(32768),
-		magnitude(0), arbdata(NULL), priorphase(0), sample(0), tone_offset(0),
+		magnitude(0), arbdata(NULL), sample(0), tone_offset(0),
 		tone_type(WAVEFORM_SINE), modulation_type(0) {
 	}
 
@@ -142,6 +143,7 @@ public:
 			freq = AUDIO_SAMPLE_RATE_EXACT / 2;
 		}
 		phase_increment = freq * (4294967296.0 / AUDIO_SAMPLE_RATE_EXACT);
+		if (phase_increment > 0x7FFE0000u) phase_increment = 0x7FFE0000;
 	}
 	void amplitude(float n) {	// 0 to 1.0
 		if (n < 0) {
@@ -198,7 +200,6 @@ private:
 	int32_t  magnitude;
 	const int16_t *arbdata;
 	uint32_t phasedata[AUDIO_BLOCK_SAMPLES];
-	uint32_t priorphase;
 	int16_t  sample; // for WAVEFORM_SAMPLE_HOLD
 	int16_t  tone_offset;
 	uint8_t  tone_type;