Remove old AudioSynthWaveform

data_waveforms.c

@@ -77,159 +77,6 @@ print "\n" unless ($len % 10) == 9;
 print "};\n";
 #endif
 
-const int16_t AudioWaveformTriangle[257] = {
-     0,   512,  1024,  1536,  2048,  2560,  3072,  3584,  4096,  4608,
-  5120,  5632,  6144,  6656,  7168,  7680,  8192,  8704,  9216,  9728,
- 10240, 10752, 11264, 11776, 12288, 12800, 13312, 13824, 14336, 14848,
- 15360, 15872, 16384, 16895, 17407, 17919, 18431, 18943, 19455, 19967,
- 20479, 20991, 21503, 22015, 22527, 23039, 23551, 24063, 24575, 25087,
- 25599, 26111, 26623, 27135, 27647, 28159, 28671, 29183, 29695, 30207,
- 30719, 31231, 31743, 32255, 32767, 32255, 31743, 31231, 30719, 30207,
- 29695, 29183, 28671, 28159, 27647, 27135, 26623, 26111, 25599, 25087,
- 24575, 24063, 23551, 23039, 22527, 22015, 21503, 20991, 20479, 19967,
- 19455, 18943, 18431, 17919, 17407, 16895, 16384, 15872, 15360, 14848,
- 14336, 13824, 13312, 12800, 12288, 11776, 11264, 10752, 10240,  9728,
-  9216,  8704,  8192,  7680,  7168,  6656,  6144,  5632,  5120,  4608,
-  4096,  3584,  3072,  2560,  2048,  1536,  1024,   512,     0,  -512,
- -1024, -1536, -2048, -2560, -3072, -3584, -4096, -4608, -5120, -5632,
- -6144, -6656, -7168, -7680, -8192, -8704, -9216, -9728,-10240,-10752,
--11264,-11776,-12288,-12800,-13312,-13824,-14336,-14848,-15360,-15872,
--16384,-16895,-17407,-17919,-18431,-18943,-19455,-19967,-20479,-20991,
--21503,-22015,-22527,-23039,-23551,-24063,-24575,-25087,-25599,-26111,
--26623,-27135,-27647,-28159,-28671,-29183,-29695,-30207,-30719,-31231,
--31743,-32255,-32767,-32255,-31743,-31231,-30719,-30207,-29695,-29183,
--28671,-28159,-27647,-27135,-26623,-26111,-25599,-25087,-24575,-24063,
--23551,-23039,-22527,-22015,-21503,-20991,-20479,-19967,-19455,-18943,
--18431,-17919,-17407,-16895,-16384,-15872,-15360,-14848,-14336,-13824,
--13312,-12800,-12288,-11776,-11264,-10752,-10240, -9728, -9216, -8704,
- -8192, -7680, -7168, -6656, -6144, -5632, -5120, -4608, -4096, -3584,
- -3072, -2560, -2048, -1536, -1024,  -512,     0
-};
-
-#if 0
-#! /usr/bin/perl
-$len = 256;
-print "const int16_t AudioWaveformTriangle[257] = {\n";
-for ($i=0; $i <= $len; $i++) {
-
-        $f = 0;
-        if ($i < $len / 4) {
-                $f = $i / ($len / 4);
-        } elsif ($i < $len * 3 / 4) {
-                $f = 2 - $i / ($len / 4);
-        } else {
-                $f = $i / ($len / 4) - 4;
-                #print "$i  $f\n";
-        }
-        $d = sprintf "%.0f", $f * 32767.0;
-        printf "%6d", $d + 0;
-        print "," if ($i < $len);
-        print "\n" if ($i % 10) == 9;
-}
-print "\n" unless ($len % 10) == 9;
-print "};\n";
-#endif
-
-const int16_t AudioWaveformSquare[257] = {
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
- 32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,-32767,
--32767,-32767,-32767,-32767,-32767,-32767, 32767
-};
-
-#if 0
-#! /usr/bin/perl
-$len = 256;
-print "const int16_t AudioWaveformSquare[257] = {\n";
-for ($i=0; $i <= $len; $i++) {
-        $f = 1.0;
-        if ($i < $len / 2) {
-                $f = 1.0;
-        } elsif ($i < $len) {
-                $f = -1.0;
-        } 
-        $d = sprintf "%.0f", $f * 32767.0;
-        printf "%6d", $d + 0;
-        print "," if ($i < $len);
-        print "\n" if ($i % 10) == 9;
-}
-print "\n" unless ($len % 10) == 9;
-print "};\n";
-#endif
-
-const int16_t AudioWaveformSawtooth[257] = {
-     0,   256,   512,   768,  1024,  1280,  1536,  1792,  2048,  2304,
-  2560,  2816,  3072,  3328,  3584,  3840,  4096,  4352,  4608,  4864,
-  5120,  5376,  5632,  5888,  6144,  6400,  6656,  6912,  7168,  7424,
-  7680,  7936,  8192,  8448,  8704,  8960,  9216,  9472,  9728,  9984,
- 10240, 10496, 10752, 11008, 11264, 11520, 11776, 12032, 12288, 12544,
- 12800, 13056, 13312, 13568, 13824, 14080, 14336, 14592, 14848, 15104,
- 15360, 15616, 15872, 16128, 16384, 16639, 16895, 17151, 17407, 17663,
- 17919, 18175, 18431, 18687, 18943, 19199, 19455, 19711, 19967, 20223,
- 20479, 20735, 20991, 21247, 21503, 21759, 22015, 22271, 22527, 22783,
- 23039, 23295, 23551, 23807, 24063, 24319, 24575, 24831, 25087, 25343,
- 25599, 25855, 26111, 26367, 26623, 26879, 27135, 27391, 27647, 27903,
- 28159, 28415, 28671, 28927, 29183, 29439, 29695, 29951, 30207, 30463,
- 30719, 30975, 31231, 31487, 31743, 31999, 32255, 32511,-32767,-32511,
--32255,-31999,-31743,-31487,-31231,-30975,-30719,-30463,-30207,-29951,
--29695,-29439,-29183,-28927,-28671,-28415,-28159,-27903,-27647,-27391,
--27135,-26879,-26623,-26367,-26111,-25855,-25599,-25343,-25087,-24831,
--24575,-24319,-24063,-23807,-23551,-23295,-23039,-22783,-22527,-22271,
--22015,-21759,-21503,-21247,-20991,-20735,-20479,-20223,-19967,-19711,
--19455,-19199,-18943,-18687,-18431,-18175,-17919,-17663,-17407,-17151,
--16895,-16639,-16384,-16128,-15872,-15616,-15360,-15104,-14848,-14592,
--14336,-14080,-13824,-13568,-13312,-13056,-12800,-12544,-12288,-12032,
--11776,-11520,-11264,-11008,-10752,-10496,-10240, -9984, -9728, -9472,
- -9216, -8960, -8704, -8448, -8192, -7936, -7680, -7424, -7168, -6912,
- -6656, -6400, -6144, -5888, -5632, -5376, -5120, -4864, -4608, -4352,
- -4096, -3840, -3584, -3328, -3072, -2816, -2560, -2304, -2048, -1792,
- -1536, -1280, -1024,  -768,  -512,  -256,     0
-};
-
-#if 0
-#! /usr/bin/perl
-$len = 256;
-print "const int16_t AudioWaveformSawtooth[257] = {\n";
-for ($i=0; $i <= $len; $i++) {
-        $f = 0;
-        if ($i < $len / 2) {
-                $f = $i / $len * 2;
-        } else {
-                $f = -2 + $i / $len * 2;
-                #print "$i  $f\n";
-        }
-        $d = sprintf "%.0f", $f * 32767.0;
-        printf "%6d", $d + 0;
-        print "," if ($i < $len);
-        print "\n" if ($i % 10) == 9;
-}
-print "\n" unless ($len % 10) == 9;
-print "};\n";
-#endif
-
-
 
 const int16_t fader_table[257] = {
     0,    1,    4,   11,   19,   30,   44,   60,   78,   99,

keywords.txt

@@ -98,10 +98,6 @@ AudioWindowWelch256	LITERAL1
 AudioWindowHamming256	LITERAL1
 AudioWindowCosine256	LITERAL1
 AudioWindowTukey256	LITERAL1
-AudioWaveformSine	LITERAL1
-AudioWaveformTriangle	LITERAL1
-AudioWaveformSquare	LITERAL1
-AudioWaveformSawtooth	LITERAL1
 
 FILTER_LOPASS	LITERAL1
 FILTER_HIPASS	LITERAL1

synth_waveform.cpp

@@ -29,87 +29,6 @@
 #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 140415 - change sin to use Paul's interpolation which is much
 //				faster than arm's sin function
@@ -270,86 +189,5 @@ void AudioSynthWaveform::update(void)
 }
 
 
-#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

@@ -30,60 +30,6 @@
 #include "AudioStream.h"
 #include "arm_math.h"
 
-//#define ORIGINAL_AUDIOSYNTHWAVEFORM
-
-#ifdef ORIGINAL_AUDIOSYNTHWAVEFORM
-// 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];
-}
-
-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
 // waveforms.c
 extern "C" {
 extern const int16_t AudioWaveformSine[257];
@@ -162,26 +108,6 @@ private:
   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