Merge pull request #28 from el-supremo/master

Fixed AudioSynthWaveform. Uses LUT with linear interpolation for sinewav...

examples/PlaySynthMusic/PlaySynthMusic.h

@@ -4,6 +4,7 @@
 //   Generated from Excel by =ROUND(2*440/32*(2^((x-9)/12)),0) for 0<x<128
 // The lowest notes might not work, depending on the Arduino clock frequency
 
+#ifdef ORIGINAL_TABLE
 const unsigned int tune_frequencies2_PGM[128] =
 {
     16,17,18,19,21,22,23,24,26,28,29,31,33,35,37,39,41,
@@ -18,7 +19,29 @@ const unsigned int tune_frequencies2_PGM[128] =
     13290,14080,14917,15804,16744,17740,18795,19912,21096,
     22351,23680,25088
 };
-
+#else
+// This is for the Teensy Audio library which specifies
+// frequencies as floating point. See make_notetab.xlsx
+const float tune_frequencies2_PGM[128] =
+{
+    8.1758,    8.6620,    9.1770,    9.7227,    10.3009,    10.9134,    11.5623,    12.2499,
+    12.9783,   13.7500,   14.5676,   15.4339,   16.3516,    17.3239,    18.3540,    19.4454,
+    20.6017,   21.8268,   23.1247,   24.4997,   25.9565,    27.5000,    29.1352,    30.8677,
+    32.7032,   34.6478,   36.7081,   38.8909,   41.2034,    43.6535,    46.2493,    48.9994,
+    51.9131,   55.0000,   58.2705,   61.7354,   65.4064,    69.2957,    73.4162,    77.7817,
+    82.4069,   87.3071,   92.4986,   97.9989,   103.8262,   110.0000,   116.5409,   123.4708,
+    130.8128,  138.5913,  146.8324,  155.5635,  164.8138,   174.6141,   184.9972,   195.9977,
+    207.6523,  220.0000,  233.0819,  246.9417,  261.6256,   277.1826,   293.6648,   311.1270,
+    329.6276,  349.2282,  369.9944,  391.9954,  415.3047,   440.0000,   466.1638,   493.8833,
+    523.2511,  554.3653,  587.3295,  622.2540,  659.2551,   698.4565,   739.9888,   783.9909,
+    830.6094,  880.0000,  932.3275,  987.7666,  1046.5023,  1108.7305,  1174.6591,  1244.5079,
+    1318.5102, 1396.9129, 1479.9777, 1567.9817, 1661.2188,  1760.0000,  1864.6550,  1975.5332,
+    2093.0045, 2217.4610, 2349.3181, 2489.0159, 2637.0205,  2793.8259,  2959.9554,  3135.9635,
+    3322.4376, 3520.0000, 3729.3101, 3951.0664, 4186.0090,  4434.9221,  4698.6363,  4978.0317,
+    5274.0409, 5587.6517, 5919.9108, 6271.9270, 6644.8752,  7040.0000,  7458.6202,  7902.1328,
+    8372.0181, 8869.8442, 9397.2726, 9956.0635, 10548.0818, 11175.3034, 11839.8215, 12543.8540
+};
+#endif
 
 #define CMD_PLAYNOTE	0x90	/* play a note: low nibble is generator #, note is next byte */
 #define CMD_STOPNOTE	0x80	/* stop a note: low nibble is generator # */

examples/PlaySynthMusic/PlaySynthMusic.ino

@@ -1,7 +1,6 @@
 // Implement the midi player inside the Audio library.
 // This uses the new version of the waveform generator code
 // See PlayMidiTones for code which uses the old version
-
  
 #include <Audio.h>
 #include <Wire.h>
@@ -38,46 +37,57 @@ AudioSynthWaveform *waves[8] = {
   &sine7,
 };
 
+// allocate a wave type to each channel.
+// The types used and their order is purely arbitrary.
+short wave_type[8] = {
+  TONE_TYPE_SINE,
+  TONE_TYPE_SQUARE,
+  TONE_TYPE_SAWTOOTH,
+  TONE_TYPE_TRIANGLE,
+  TONE_TYPE_SINE,
+  TONE_TYPE_SQUARE,
+  TONE_TYPE_SAWTOOTH,
+  TONE_TYPE_TRIANGLE,
+};
+
+// Two mixers are needed to handle the 8 channels of music
 AudioMixer4              mixer1;
 AudioMixer4              mixer2;
 AudioOutputI2S audioOut;
 
+// Mix the first four channels into mixer1
 AudioConnection c0(sine0, 0, mixer1, 0);
 AudioConnection c1(sine1, 0, mixer1, 1);
 AudioConnection c2(sine2, 0, mixer1, 2);
 AudioConnection c3(sine3, 0, mixer1, 3);
 
+// and the last 4 channels into mixer2
 AudioConnection c4(sine4, 0, mixer2, 0);
 AudioConnection c5(sine5, 0, mixer2, 1);
 AudioConnection c6(sine6, 0, mixer2, 2);
 AudioConnection c7(sine7, 0, mixer2, 3);
+
+// Output mixre1 to the left channel and mixer2 to the right
 AudioConnection c11(mixer1, 0, audioOut, 0);
 AudioConnection c12(mixer2, 0, audioOut, 1);
 
 //AudioControl_WM8731 codec;
 AudioControlSGTL5000 codec;
 
+// Initial value of the volume control
 int volume = 50;
 
-// allocate a wave type to each channel.
-// The types used and their order is purely arbitrary.
-short wave_type[8] = {
-  TONE_TYPE_SINE,
-  TONE_TYPE_SQUARE,
-  TONE_TYPE_SAWTOOTH,
-  TONE_TYPE_TRIANGLE,
-  TONE_TYPE_SINE,
-  TONE_TYPE_SQUARE,
-  TONE_TYPE_SAWTOOTH,
-  TONE_TYPE_TRIANGLE,
-};
-
 void setup()
 {
   Serial.begin(115200);
   while (!Serial) ;
-  delay(3000);
-
+  delay(2000);
+  
+// http://gcc.gnu.org/onlinedocs/cpp/Standard-Predefined-Macros.html
+  Serial.print("Begin ");
+  Serial.println(__FILE__);
+  
+  // Proc = 12 (13),  Mem = 2 (8)
   // Audio connections require memory to work.
   // The memory usage code indicates that 8 is the maximum
   // so give it 10 just to be sure.
@@ -89,14 +99,11 @@ void setup()
   // Comment this if you don't it
   codec.unmuteLineout();
 
-  
   // Set the ramp time for each wave object
   for(int i = 0; i < 8;i++) {
     waves[i]->set_ramp_length(88);
   }
 
-  Serial.println("Begin PlayMidiTones");
-
   Serial.println("setup done");
   
   // Initialize processor and memory measurements
@@ -111,8 +118,8 @@ void loop()
   unsigned char c,opcode,chan;
   unsigned long d_time;
   
-// Change this to if(1) for measurement output
-if(0) {
+// Change this to if(1) for measurement output every 5 seconds
+if(1) {
 /*
   For PlaySynthMusic this produces:
   Proc = 20 (21),  Mem = 2 (8)
@@ -138,6 +145,7 @@ if(0) {
     codec.volume((float)n / 10.23);
   }
   
+  // read the next note from the table
   c = *sp++;
   opcode = c & 0xf0;
   // was 0x0f but I'm only handling 8 channels
@@ -163,7 +171,8 @@ if(0) {
     return;
   }
   
-  // Play the note on 'chan'
+  // Play the note on 'chan' - divide the frequency by two because the
+  // table of frequencies has been doubled.
   if(opcode == CMD_PLAYNOTE) {
     waves[chan]->begin(AMPLITUDE,tune_frequencies2_PGM[*sp++],
                         wave_type[chan]);

examples/fir_filter/fir_filter.ino

@@ -1,4 +1,8 @@
 /*
+c
+  - released
+b
+- Use FIR filters with fast_fft option
 
 The audio board uses the following pins.
  6 - MEMCS
@@ -25,14 +29,15 @@ The audio board uses the following pins.
 #include <Bounce.h>
 #include "filters.h"
 
-// If this pin is grounded the FIR filter is turned which
-// makes just pass through the audio
+// If this pin is grounded the FIR filter is turned off
+// which just passes the audio sraight through
 // Don't use any of the pins listed above
 #define PASSTHRU_PIN 1
 // If this pin goes low the next FIR filter in the list
 // is switched in.
 #define FILTER_PIN 0
 
+// debounce the passthru and filter switching pins
 Bounce b_passthru = Bounce(PASSTHRU_PIN,15);
 Bounce b_filter   = Bounce(FILTER_PIN,15);
 
@@ -41,16 +46,18 @@ const int myInput = AUDIO_INPUT_LINEIN;
 
 
 AudioInputI2S       audioInput;         // audio shield: mic or line-in
-AudioFilterFIR           myFilter;
+// Use the fast FIR filter for left and right channels
+AudioFilterFIR       myFilterL(true);
+AudioFilterFIR       myFilterR(true);
 AudioOutputI2S      audioOutput;        // audio shield: headphones & line-out
 
 // Create Audio connections between the components
-// Both channels of the audio input go to the FIR filter
-AudioConnection c1(audioInput, 0, myFilter, 0);
-AudioConnection c2(audioInput, 1, myFilter, 1);
-// both channels from the FIR filter go to the audio output
-AudioConnection c3(myFilter, 0, audioOutput, 0);
-AudioConnection c4(myFilter, 1, audioOutput, 1);
+// Route audio into the left and right filters
+AudioConnection c1(audioInput, 0, myFilterL, 0);
+AudioConnection c2(audioInput, 1, myFilterR, 0);
+// Route the output of the filters to their respective channels
+AudioConnection c3(myFilterL, 0, audioOutput, 0);
+AudioConnection c4(myFilterR, 0, audioOutput, 1);
 AudioControlSGTL5000 audioShield;
 
 
@@ -62,9 +69,9 @@ struct fir_filter {
 // index of current filter. Start with the low pass.
 int fir_idx = 0;
 struct fir_filter fir_list[] = {
-  low_pass , 100,    // low pass with cutoff at 1kHz and -60dB at 2kHz
-  band_pass, 100,    // bandpass 1200Hz - 1700Hz
-  NULL,      0
+  {low_pass , 100},    // low pass with cutoff at 1kHz and -60dB at 2kHz
+  {band_pass, 100},    // bandpass 1200Hz - 1700Hz
+  {NULL,      0}
 };
 
 
@@ -98,8 +105,8 @@ void setup() {
     Serial.println(") is grounded");
   }  
   // Initialize the filter
-  myFilter.begin(fir_list[0].coeffs,fir_list[0].num_coeffs);
-  
+  myFilterL.begin(fir_list[0].coeffs,fir_list[0].num_coeffs);
+  myFilterR.begin(fir_list[0].coeffs,fir_list[0].num_coeffs);  
   Serial.println("setup done");
 }
 
@@ -108,7 +115,7 @@ int old_idx = -1;
 
 // audio volume
 int volume = 0;
-
+unsigned long last_time = millis();
 void loop()
 {
   // Volume control
@@ -122,18 +129,20 @@ void loop()
   b_passthru.update();
   b_filter.update();
   
-
-  
   // If the passthru button is pushed, save the current
   // filter index and then switch the filter to passthru
   if(b_passthru.fallingEdge()) {
     old_idx = fir_idx;
-    myFilter.begin(FIR_PASSTHRU,0);
+    myFilterL.begin(FIR_PASSTHRU,0);
+    myFilterR.begin(FIR_PASSTHRU,0);
   }
   
   // If passthru button is released, restore previous filter
   if(b_passthru.risingEdge()) {
-    if(old_idx != -1)myFilter.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+    if(old_idx != -1) {
+      myFilterL.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+      myFilterR.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+    }
     old_idx = -1;
   }
   
@@ -141,9 +150,31 @@ void loop()
   if(b_filter.fallingEdge()) {
     fir_idx++;
     if(fir_list[fir_idx].num_coeffs == 0)fir_idx = 0;
-    myFilter.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+    myFilterL.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+    myFilterR.begin(fir_list[fir_idx].coeffs,fir_list[fir_idx].num_coeffs);
+  }
+
+
+if(1) {
+// With fast_fir
+// Proc = 18 (18),  Mem = 4 (6)
+
+  if(millis() - last_time >= 5000) {
+    Serial.print("Proc = ");
+    Serial.print(AudioProcessorUsage());
+    Serial.print(" (");    
+    Serial.print(AudioProcessorUsageMax());
+    Serial.print("),  Mem = ");
+    Serial.print(AudioMemoryUsage());
+    Serial.print(" (");    
+    Serial.print(AudioMemoryUsageMax());
+    Serial.println(")");
+    last_time = millis();
   }
 }
 
 
+}
+
+
 

filter_fir.cpp

@@ -29,7 +29,6 @@ void AudioFilterFIR::begin(short *cp,int n_coeffs)
   // Initialize FIR instances for the left and right channels
   if(coeff_p && (coeff_p != FIR_PASSTHRU)) {
     arm_fir_init_q15(&l_fir_inst, n_coeffs, coeff_p, &l_StateQ15[0], AUDIO_BLOCK_SAMPLES);
-    arm_fir_init_q15(&r_fir_inst, n_coeffs, coeff_p, &r_StateQ15[0], AUDIO_BLOCK_SAMPLES);
   }
 }
 
@@ -55,11 +54,6 @@ void AudioFilterFIR::update(void)
       transmit(block,0);
       release(block);
     }
-    block = receiveReadOnly(1);
-    if(block) {
-      transmit(block,1);
-      release(block);
-    }
     return;
   }
   // Left Channel
@@ -67,22 +61,15 @@ void AudioFilterFIR::update(void)
   // get a block for the FIR output
   b_new = allocate();
   if(block && b_new) {
-    arm_fir_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
+    if(arm_fast)
+    	arm_fir_fast_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
+    else
+    	arm_fir_q15(&l_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES); 
     // send the FIR output to the left channel
     transmit(b_new,0);
   }
   if(block)release(block);
   if(b_new)release(b_new);
-
-  // Right Channel
-  block = receiveReadOnly(1);
-  b_new = allocate();
-  if(block && b_new) {
-    arm_fir_q15(&r_fir_inst, (q15_t *)block->data, (q15_t *)b_new->data, AUDIO_BLOCK_SAMPLES);
-    transmit(b_new,1);
-  }
-  if(block)release(block);
-  if(b_new)release(b_new);
 }
 
 

filter_fir.h

@@ -26,6 +26,10 @@
 #include "AudioStream.h"
 #include "arm_math.h"
 
+#define USE_FAST_FIR true
+#define USE_SLOW_FIR false
+
+
 // Maximum number of coefficients in a FIR filter
 // The audio breaks up with 128 coefficients so a
 // maximum of 150 is more than sufficient
@@ -39,12 +43,12 @@ class AudioFilterFIR :
 public AudioStream
 {
 public:
-  AudioFilterFIR(void): 
-  AudioStream(2,inputQueueArray), coeff_p(NULL)
+  AudioFilterFIR(const boolean a_f): 
+  AudioStream(2,inputQueueArray), arm_fast(a_f), coeff_p(NULL)
   { 
   }
 
-  void begin(short *coeff_p,int f_pin);
+  void begin(short *coeff_p,int n_coeffs);
   virtual void update(void);
   void stop(void);
   
@@ -54,11 +58,11 @@ private:
   // the state arrays are defined to handle a maximum of MAX_COEFFS
   // coefficients in a filter
   q15_t l_StateQ15[AUDIO_BLOCK_SAMPLES + MAX_COEFFS];
-  q15_t r_StateQ15[AUDIO_BLOCK_SAMPLES + MAX_COEFFS];
   arm_fir_instance_q15 l_fir_inst;
-  arm_fir_instance_q15 r_fir_inst;
   // pointer to current coefficients or NULL or FIR_PASSTHRU
   short *coeff_p;
+  // Whether to use the fast arm FIR code 
+  const boolean arm_fast;
 };
 
 #endif

filter_fir.md

@@ -0,0 +1,18 @@
+filter_fir 140418
+	Filters the audio stream using FIR coefficients supplied by the user.
+	The ARM	library has two q15 functions which perform an FIR filter. One uses
+	a 64-bit accumulator (arm_fir_q15) and the other uses a 32-bit accumulator
+	(arm_fir_fast_q15). When instantiating a filter a boolean argument specifies
+	which version to use. Specifying USE_FAST_FIR (defined as boolean true) uses
+	the fast code otherwise the slower code is used. For example:
+	AudioFilterFIR       myFilterL(USE_FAST_FIR);
+
+void begin(short *cp,int n_coeffs)
+	Starts the filter using coefficients at *cp and the number of coefficients
+	is n_coeffs. The special value FIR_PASSTHRU can be used in place of the
+	address of the coefficient array, in which case the function will just pass
+	audio samples through without filtering.
+
+void stop(void)
+	Stops the filter.
+

synth_waveform.cpp

@@ -29,9 +29,95 @@
 #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
+// PAH 140316 - fix calculation of sample (amplitude error)
+// PAH 140314 - change t_hi from int to float
+// PAH - add ramp-up and ramp-down to the onset of the wave
+// the length is specified in samples
+
 void AudioSynthWaveform::set_ramp_length(int16_t r_length)
 {
   if(r_length < 0) {
@@ -47,17 +133,15 @@ void AudioSynthWaveform::set_ramp_length(int16_t r_length)
 }
 
 
-boolean AudioSynthWaveform::begin(float t_amp,int t_hi,short type)
+boolean AudioSynthWaveform::begin(float t_amp,float t_hi,short type)
 {
   tone_type = type;
-//  tone_amp = t_amp;
   amplitude(t_amp);
-  tone_freq = t_hi;
-  if(t_hi < 1)return false;
+  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;
-//  tone_incr = (0x100000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
-  tone_incr = (0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
+  frequency(t_hi);
   if(0) {
     Serial.print("AudioSynthWaveform.begin(tone_amp = ");
     Serial.print(t_amp);
@@ -72,8 +156,6 @@ 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)
 {
@@ -81,10 +163,12 @@ void AudioSynthWaveform::update(void)
   short *bp;
   // temporary for ramp in sine
   uint32_t ramp_mag;
+  int32_t val1, val2, val3;
+  uint32_t index, scale;
+  
   // temporaries for TRIANGLE
   uint32_t mag;
   short tmp_amp;
-
   
   if(tone_freq == 0)return;
   //          L E F T  C H A N N E L  O N L Y
@@ -94,6 +178,14 @@ void AudioSynthWaveform::update(void)
     switch(tone_type) {
     case TONE_TYPE_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;
         // 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
@@ -106,8 +198,8 @@ void AudioSynthWaveform::update(void)
           ramp_up--;
           // adjust tone_phase to Q15 format and then adjust the result
           // of the multiplication
-      	// calculate the sample
-          tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
+      	  // calculate the sample
+          tmp_amp = (short)((val3 * tone_amp) >> 15);
           *bp++ = (tmp_amp * ramp_mag)>>15;
         } 
         else if(ramp_down) {
@@ -120,15 +212,10 @@ void AudioSynthWaveform::update(void)
           // 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
-          tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * last_tone_amp) >> 17);
+		  tmp_amp = (short)((val3 * last_tone_amp) >> 15);
           *bp++ = (tmp_amp * ramp_mag)>>15;
         } else {
-          // adjust tone_phase to Q15 format and then adjust the result
-          // of the multiplication
-          tmp_amp = (short)((arm_sin_q15(tone_phase>>16) * tone_amp) >> 17);
-          *bp++ = tmp_amp;
+		  *bp++ = (short)((val3 * tone_amp) >> 15);
         } 
         
         // phase and incr are both unsigned 32-bit fractions
@@ -140,7 +227,7 @@ void AudioSynthWaveform::update(void)
       
     case TONE_TYPE_SQUARE:
       for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        if(tone_phase & 0x80000000)*bp++ = -tone_amp;
+        if(tone_phase & 0x40000000)*bp++ = -tone_amp;
         else *bp++ = tone_amp;
         // phase and incr are both unsigned 32-bit fractions
         tone_phase += tone_incr;
@@ -149,10 +236,9 @@ void AudioSynthWaveform::update(void)
       
     case TONE_TYPE_SAWTOOTH:
       for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
-        *bp = ((short)(tone_phase>>16)*tone_amp) >> 15;
-        bp++;
+        *bp++ = ((short)(tone_phase>>15)*tone_amp) >> 15;
         // phase and incr are both unsigned 32-bit fractions
-        tone_phase += tone_incr;
+        tone_phase += tone_incr;    
       }
       break;
 
@@ -173,7 +259,7 @@ void AudioSynthWaveform::update(void)
           mag = ~mag + 1;
         }
         *bp++ = ((short)(mag>>17)*tmp_amp) >> 15;
-        tone_phase += tone_incr;
+        tone_phase += 2*tone_incr;
       }
       break;
     }
@@ -182,3 +268,88 @@ 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

@@ -30,6 +30,64 @@
 #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];
+}
 
 #define AUDIO_SAMPLE_RATE_ROUNDED (44118)
 
@@ -51,9 +109,10 @@ public:
   { 
   }
   
-  void frequency(int t_hi)
+  void frequency(float t_hi)
   {
-    tone_incr = (0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT;
+    if (t_hi > AUDIO_SAMPLE_RATE_EXACT / 2 || t_hi < 0.0) return;
+    tone_incr = ((0x80000000LL*t_hi)/AUDIO_SAMPLE_RATE_EXACT) + 0.5;
   }
   
   // If ramp_length is non-zero this will set up
@@ -85,7 +144,7 @@ public:
     tone_amp = n * 32767.0;
   }
   
-  boolean begin(float t_amp,int t_hi,short t_type);
+  boolean begin(float t_amp,float t_hi,short t_type);
   virtual void update(void);
   void set_ramp_length(int16_t r_length);
   
@@ -102,4 +161,27 @@ 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

synth_waveform.md

@@ -0,0 +1,36 @@
+synth_waveform 140404
+	This synthesizes a waveform of the specified type with a given amplitude
+	and frequency. There are currently four types of waveform:
+	#define TONE_TYPE_SINE     0
+	#define TONE_TYPE_SAWTOOTH 1
+	#define TONE_TYPE_SQUARE   2
+	#define TONE_TYPE_TRIANGLE 3
+	Sine wave generation uses a lookup table and linear interpolation.
+	The other three waveforms are generated directly without using table lookup.
+	
+boolean begin(float t_amp,float t_freq,short t_type)
+	This starts generation of a waveform of given type, amplitude and frequency
+	Example: begin(0.8,440.0,TONE_TYPE_SINE)
+	
+void set_ramp_length(int16_t r_length)
+	When a tone starts, or ends, playing it can generate an audible "thump" which can
+	be very distracting, especially when playing musical notes. This function specifies
+	a "ramp" length (in number of samples) and the beginning of the generated waveform
+	will be ramped up in volume from zero to t_amp over the course of r_length samples.
+	When the tone is switched off, by changing its volume to zero, instead of ending
+	abruptly it will be ramped down to zero over the next r_length samples. 
+	For example, if r_length is 44, the beginning and end of the wave will have a ramp
+	of approximately one millisecond.
+	
+void frequency(float t_freq)
+	Changes the frequency of the wave to the specified t_freq. This is done in a phase-
+	continuous manner which should allow generation of audio frequency shift keying and
+	other effects requiring a changing frequency.
+	If the frequency is set to zero sample generation is stopped.
+	
+void amplitude(float n)
+	Changes the amplitude to 'n'. If 'n' is zero the wave is turned off and any further
+	audio output will be zero.
+
+
+