Showing posts with label Audio. Show all posts
Showing posts with label Audio. Show all posts

Saturday, December 1, 2012

Arduino Due DDS - Part 1 - Sinewaves and Fixed Point Maths

This post provides a quick introduction to Direct Digital Synthesis (DDS) on the Arduino Due. The sample sketch outputs a sinewave at a 44.1Khz sample rate with 12 bit resolution using the built in Digital To Analog Converter (DAC).

A related sketch which provides more interesting audio output without the explanation can be found here -
http://rcarduino.blogspot.com/2012/11/quick-and-dirty-synth-for-arduino-due.html

All of the Audio projects on RCArduino are based on this same DDS technique described below, see the following links for DDS Projects -
http://rcarduino.blogspot.com/2012/10/five-dollar-synthesiser.html
http://rcarduino.blogspot.com/2012/08/the-must-build-arduino-project-illutron.html

http://rcarduino.blogspot.com/2012/11/auduino-with-delay.html
http://rcarduino.blogspot.com/2012/08/adding-audio-to-arduino-projects.html
http://rcarduino.blogspot.com/2012/10/arduino-modular-synthesizer-part-one.html

Direct Digital Synthesis - Introduction
DDS is a common approach to generating sinewave output from digital systems. The approach is based around outputting samples from a description of the required waveform held in memory. The description is an array of samples which can be traversed to plot the waveform onscreen or by using DDS can generate a sinewave output at a selected frequency.

The 600 sample 12 Bit sine wave table used in the sketch below plotted on screen using processing -

The 128 sample 8 Bit wave tables used in the Illutron B synth -


Direct Digital Synthesis - Sample Rate
To generate an output waveform we need a way to regularly update the output value, we do this by using a timer interrupt. The frequency of this interrupt is termed the sample rate, in our case we are using a 44.1Khz sample rate meaning that we have an interrupt triggering 44,100 times per second.

As this interrupt is triggering at such a high rate it is very important that we keep it efficient, this is the reason for using a pre computed wavetable rather than computing the sine samples on the fly.

Direct Digital Synthesis - Frequency Generation
We can generate different frequencies by changing the rate at which we traverse through the wave table. One pass through the wavetable provides one complete cycle of the selected waveform (sine in our case). If we pass through the table once in 44,100 interrupts we generate a frequency of 1Hz, if we pass through 100 times faster, we get a sinewave output at a frequency of 100Hz.

Direct Digital Synthesis - Implementation and Terminology
The following terms are commonly used in reference to DDS and variations are used throughout the RC Arduino audio projects.

1) Phase Accumulator - this is an array index which points to the current sample in the selected wavetable.
2) Phase Increment - this value is used to advance the Phase Accumulator each time the interrupt is called.

Example usage of the phase accumulator used inside the timer interrupt to select the next sample in the wave table array -

// Update the phase accumulator by adding the phase increment
ulPhaseAccumulator += ulPhaseIncrement;

// get the current sample from the sine table using the phase accumulator as the index  
uint32_t ulOutput = nSineTable[ulPhaseAccumulator>>20];

Direct Digital Synthesis - Fixed Point Maths
To generate a full range of frequencies we will often need to use fractional values for our phase accumulator.


The usual method of working with fractional values is to use the float data type however in DDS we need our interrupts to run as fast as possible. With very few exceptions microcontrollers are not able to process the float data type at high speed, the solution is to use fixed point integer maths. 

Fixed Points For Higher Integer Precision
There are many applications of fixed point arithmetic but in DDS the main use is to provide increased precision - equivalent to having many decimal points while still using the high speed integer (whole number) data type.

To get extra precision we use more bits than we need to represent our data range. For example the sample code below has a wave table size of 600 samples so the range of we need for our phase accumulator is 0 to 599.

We can represent this range with 10 bits but if we use 16 bits we have 6 additional bits of precession.


16 Bit Integer














Bit  15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0
Value 32768 16384 8192 4096 2048 1024 512 256 128 64 32 16 8 4 2 1

















Fixed Point 10.6 using 16 Bit Integer












Bit  15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0
Value 1024 512 256 128 64 32 16 8 4 2 1 0.5 0.25 0.125 0.0625 0.0313

We use this additional precision to perform fractional addition to the phase accumulator, the 10.6 format gives 1/32 precision as the least significant bit.

Fixed point addition - we know its fixed point, the compiler doesn't

// Update the phase accumulator by adding the phase increment
ulPhaseAccumulator += ulPhaseIncrement;  

This line of code is actually adding two fixed point numbers, part of the trick is that the compiler does not know this, it sees them simply as two integers which it processes using very fast hardware integer addition.

When we want to use the phase accumulator as the array index, we perform an arithmetic shift six places to the right, this drops off the six bits of additional precession and leaves us with a 10 bit integer value for the array index.

// drop the six bits of additional precision and access the remaining 10 bits as an integer array index in the range 0 - 599

uint16_t unOutputValue = sineTable[ulPhaseAccumulator>>6];

The only trick to fixed point maths is to figure out a scheme which will work for your project and stick to it. Remember the compiler does not know that you are treating part of the number as fractions so the responsibility is on you the programmer to keep consistency.

Fixed Point In The RCArduino DDS Example Sketch
The standard data type for the 32 bit Arduino Due is a 32 bit integer, this gives us scope for very high levels of precession in integer maths. The scheme used for the phase increment and phase accumulator variables in the sample sketch is 12.20 (12+20 = 32 Bits) this can be seen in the following line of code where the precision is shifted out to access the pure 12 bit integer value -

uint32_t ulOutput = nSineTable[ulPhaseAccumulator>>20];

Where ever you see code that makes extensive use of '>>' operations you might be looking at fixed point arithmetic, in a follow up post we will look at how we perform fast fixed point multiplication. This is another key to fast and flexible synthesizer engines.

Fixed Point Multiplication is used in Audio synthesis to apply envelopes and amplitude modulation.

Envelope Applied to a DDS sine wave using Fixed Point Maths in the Illutron B Project -

DDS Sinewave for Arduino Due - The Sketch
Use a potentiometer connected to analog pin 0 to control the pitch. To hear the output you can use an amplifier circuit such as this one used throughout RCArduino - 

http://rcarduino.blogspot.com/2012/08/adding-audio-to-arduino-projects.html

The sinewave is output through DAC0 - Note that the DAC does not provide a full 0-3.3volt swing, this is a hardware limitation of the SAM3X8E chip.


Caution : The SAM3X8E microcontroller at the heart of the Arduino Due is less able to sink and source current than the AVR family of chips used in the 8-Bit Arduinos. See the update below for the most recent recommendations - 

Update 09/06/2013 - Gaétan Ro has taken things much further in the latest build of his Groovuino project. The Project uses the technique described here combined with some sensors, a sequencer and filter to produce a really nice sounding groovebox. I plan to build one of these for myself and hope to be able to help Gaétan with its further development.

http://www.youtube.com/watch?v=EqI2hEVbMPI

Gaéta's blog -  http://groovuino.blogspot.com/

Update 28/03/2013 : There is little information on the SAM3X8E DAC in the datasheet however Arduino forum used 'stimmer' has found recommendations on a closely related chip which suggest that a 2K Ohm resistor should be placed as a series current limiting resistor between the DAC Output and the load. This assumes that the load has no resistance of its own which will be the case if you accidentally short something, if you know the resistance of your load, you can reduce this value, if not, its a reasonable starting point and will protect against accidents. For more information see the original thread here - http://arduino.cc/forum/index.php/topic,139733.15.html

Update 28/03/2013 : Gaétan Ro has been building on the techniques here to produce an Arduino Due based groove box he is calling the 'Groovuino'. You can hear a clip of the project in action here -

https://soundcloud.com/gaetino/groovuino

And follow the future development on the blog Gaétan has started here -

http://groovuino.blogspot.ae/

// RCArduino DDS Sinewave for Arduino Due
// RCArduino DDS Sinewave by RCArduino is licensed under a Creative Commons Attribution 3.0 Unported License.
// Based on a work at rcarduino.blogspot.com.

// For helpful background information on Arduino Due Timer Configuration, refer to the following link
// thanks to Sebastian Vik
// http://arduino.cc/forum/index.php?action=post;topic=130423.15;num_replies=20

// For background information on the DDS Technique see
// http://interface.khm.de/index.php/lab/experiments/arduino-dds-sinewave-generator/

// For audio sketches making extensive use of DDS Techniques, search the RCArduino Blog
// for the tags Audio or synth

// These are the clock frequencies available to the timers /2,/8,/32,/128
// 84Mhz/2 = 42.000 MHz
// 84Mhz/8 = 10.500 MHz
// 84Mhz/32 = 2.625 MHz
// 84Mhz/128 = 656.250 KHz
//
// 44.1Khz = CD Sample Rate
// Lets aim for as close to the CD Sample Rate as we can get -
//
// 42Mhz/44.1Khz = 952.38
// 10.5Mhz/44.1Khz = 238.09 // best fit divide by 8 = TIMER_CLOCK2 and 238 ticks per sample
// 2.625Hmz/44.1Khz = 59.5
// 656Khz/44.1Khz = 14.88

// 84Mhz/44.1Khz = 1904 instructions per tick

// the phase accumulator points to the current sample in our wavetable
uint32_t ulPhaseAccumulator = 0;
// the phase increment controls the rate at which we move through the wave table
// higher values = higher frequencies
volatile uint32_t ulPhaseIncrement = 0;   // 32 bit phase increment, see below

// full waveform = 0 to SAMPLES_PER_CYCLE
// Phase Increment for 1 Hz =(SAMPLES_PER_CYCLE_FIXEDPOINT/SAMPLE_RATE) = 1Hz
// Phase Increment for frequency F = (SAMPLES_PER_CYCLE/SAMPLE_RATE)*F
#define SAMPLE_RATE 44100.0
#define SAMPLES_PER_CYCLE 600
#define SAMPLES_PER_CYCLE_FIXEDPOINT (SAMPLES_PER_CYCLE<<20)
#define TICKS_PER_CYCLE (float)((float)SAMPLES_PER_CYCLE_FIXEDPOINT/(float)SAMPLE_RATE)

// to represent 600 we need 10 bits
// Our fixed point format will be 10P22 = 32 bits


// We have 521K flash and 96K ram to play with

// Create a table to hold the phase increments we need to generate midi note frequencies at our 44.1Khz sample rate
#define MIDI_NOTES 128
uint32_t nMidiPhaseIncrement[MIDI_NOTES];

// fill the note table with the phase increment values we require to generate the note
void createNoteTable(float fSampleRate)
{
  for(uint32_t unMidiNote = 0;unMidiNote < MIDI_NOTES;unMidiNote++)
  {
    // Correct calculation for frequency
    Serial.print(unMidiNote);
    Serial.print(" ");
    float fFrequency = ((pow(2.0,(unMidiNote-69.0)/12.0)) * 440.0);
    Serial.print(fFrequency);
    Serial.print(" ");
   
    nMidiPhaseIncrement[unMidiNote] = fFrequency*TICKS_PER_CYCLE;
    Serial.println(nMidiPhaseIncrement[unMidiNote]);
  }
}

// Create a table to hold pre computed sinewave, the table has a resolution of 600 samples
#define WAVE_SAMPLES 600
// default int is 32 bit, in most cases its best to use uint32_t but for large arrays its better to use smaller
// data types if possible, here we are storing 12 bit samples in 16 bit ints
uint16_t nSineTable[WAVE_SAMPLES];

// create the individual samples for our sinewave table
void createSineTable()
{
  for(uint32_t nIndex = 0;nIndex < WAVE_SAMPLES;nIndex++)
  {
    // normalised to 12 bit range 0-4095
    nSineTable[nIndex] = (uint16_t)  (((1+sin(((2.0*PI)/WAVE_SAMPLES)*nIndex))*4095.0)/2);
    Serial.println(nSineTable[nIndex]);
  }
}

void setup()
{
  Serial.begin(9600);

  createNoteTable(SAMPLE_RATE);
  createSineTable();
 
  /* turn on the timer clock in the power management controller */
  pmc_set_writeprotect(false);
  pmc_enable_periph_clk(ID_TC4);

  /* we want wavesel 01 with RC */
  TC_Configure(/* clock */TC1,/* channel */1, TC_CMR_WAVE | TC_CMR_WAVSEL_UP_RC | TC_CMR_TCCLKS_TIMER_CLOCK2);
  TC_SetRC(TC1, 1, 238); // sets <> 44.1 Khz interrupt rate
  TC_Start(TC1, 1);
 
  // enable timer interrupts on the timer
  TC1->TC_CHANNEL[1].TC_IER=TC_IER_CPCS;
  TC1->TC_CHANNEL[1].TC_IDR=~TC_IER_CPCS;
 
  /* Enable the interrupt in the nested vector interrupt controller */
  /* TC4_IRQn where 4 is the timer number * timer channels (3) + the channel number (=(1*3)+1) for timer1 channel1 */
  NVIC_EnableIRQ(TC4_IRQn);

  // this is a cheat - enable the DAC
  analogWrite(DAC0,0);
}

void loop()
{
  // read analog input 0 drop the range from 0-1024 to 0-127 with a right shift 3 places,
  // then look up the phaseIncrement required to generate the note in our nMidiPhaseIncrement table
  uint32_t ulInput = analogRead(0);
  ulPhaseIncrement = nMidiPhaseIncrement[ulInput>>3];
}

void TC4_Handler()
{
  // We need to get the status to clear it and allow the interrupt to fire again
  TC_GetStatus(TC1, 1);
 
  ulPhaseAccumulator += ulPhaseIncrement;   // 32 bit phase increment, see below

  // if the phase accumulator over flows - we have been through one cycle at the current pitch,
  // now we need to reset the grains ready for our next cycle
  if(ulPhaseAccumulator > SAMPLES_PER_CYCLE_FIXEDPOINT)
  {
   // DB 02/Jan/2012 - carry the remainder of the phase accumulator
   ulPhaseAccumulator -= SAMPLES_PER_CYCLE_FIXEDPOINT;
   }

  // get the current sample  
  uint32_t ulOutput = nSineTable[ulPhaseAccumulator>>20];
 
  // we cheated and user analogWrite to enable the dac, but here we want to be fast so
  // write directly 
  dacc_write_conversion_data(DACC_INTERFACE, ulOutput);
}

Friday, November 30, 2012

Quick And Dirty Synth For Arduino Due

The first RCArduino project for the Arduino Due is the Quick And Dirty Synth. The synth is a simple showcase for audio output through the DAC running at the Audio CD sample rate of 44.1Khz.


Creative Commons License
RCArduino Quick And Dirty Synth by RCArduino is licensed under a
Creative Commons Attribution 3.0 Unported License.
Based on a work at rcarduino.blogspot.com.


The showcase is based on the simplest synth engine I could create - 3 counters counting up at a rate controlled by three analog inputs.

It might sound simple but its a surprisingly rich sounding synth engine, here is how it works - 

Two of the counters generate ramp waveforms - think about it, they are counting up from 0 to 4095, when they overflow, they go back to 0 and start the count again.


Example Ramp Output -  




Example Ramp At A Higher Frequency -



These ramp waveforms are summed together at the output to generate a more complex waveform - two ramp waves of independent frequency superimposed on each other.

Example - Two triangle at frequency F and 3F mixed together to create a new output waveform.

The third ramp waveform is used to control the pitch. It is not mixed with the output waveforms instead it achieves pitch control by resetting the first two waveforms.


Output waveform reset at frequency determined by third counter - notice that the counter is not directly present in the output but controls the repetition/synchronization of the output which in turn creates the pitch. 


To hear the technique in action connect three potentiometers to your Arduino Due on analog inputs 0,1,2 and an audio amplifier to DAC0.

All of the Audio projects on RCArduino use the following simple amplifier circuit -
http://rcarduino.blogspot.com/2012/08/adding-audio-to-arduino-projects.html

To learn more about the synthesis technique used in the quick and dirty synthesizer see the following link -

http://rcarduino.blogspot.com/2012/08/adding-audio-to-arduino-projects.html

The code 

Caution : The SAM3X8E microcontroller at the heart of the Arduino Due is less able to sink and source current than the AVR family of chips used in the 8-Bit Arduinos. I would suggest using a series resistor of around 500 Ohms when connecting the Arduino Due DAC to an external device. See this thread on the Arduino forum for more information - http://arduino.cc/forum/index.php/topic,139733.0.html
 
// RCArduino Quick And Dirty Synth for Arduino Due// RCArduino Quick And Dirty Synth by RCArduino is licensed under a Creative Commons Attribution 3.0 Unported License.
// Based on a work at rcarduino.blogspot.com.

// For helpful background information on Arduino Due Timer Configuration, refer to the following link
// thanks to Sebastian Vik
// http://arduino.cc/forum/index.php?action=post;topic=130423.15;num_replies=20

// The following folders within the arduino install provide access to source code and documentation
// its very low level, the documentation is next to useless, but it might help someone
// C:\arduino-1.5.1r2\hardware\arduino\sam\system\libsam\source
// C:/arduino-1.5.1r2/hardware/arduino/sam/system/CMSIS/Device/ATMEL/sam3xa/html/tc1.html
// C:/arduino-1.5.1r2/hardware/arduino/sam/system/CMSIS/Device/ATMEL/sam3xa/html/SAM3XA.html
// C:\arduino-1.5.1r2\hardware\arduino\sam\system\CMSIS\CMSIS\Documentation

// These are the clock frequencies available to the timers /2,/8,/32,/128
// 84Mhz/2 = 42.000 MHz
// 84Mhz/8 = 10.500 MHz
// 84Mhz/32 = 2.625 MHz
// 84Mhz/128 = 656.250 KHz
//
// 44.1Khz = CD Sample Rate
// Lets aim for as close to the CD Sample Rate as we can get -
//
// 42Mhz/44.1Khz = 952.38
// 10.5Mhz/44.1Khz = 238.09 // best fit divide by 8 = TIMER_CLOCK2 and 238 ticks per sample
// 2.625Hmz/44.1Khz = 59.5
// 656Khz/44.1Khz = 14.88

// 84Mhz/44.1Khz = 1904 instructions per tick

// These variables represent our synth engine, its similar to a vastly simplified Auduino and has a similar sound.
// Phase Accumulator controls the main pitch, grain1 and grain two phase accumulator control the pitch for the two grains
// The grainPhaseAccumulators are essentially simple counters, when you count up and overflow back to 0, you generate a ramp waveform
// we generate two and mix them together.
uint32_t ulPhaseAccumulator = 0; // 32 bit phase accumulator, if we shift >> 20 bits we get a 12 bit value for our output ADC
volatile uint32_t ulPhaseIncrement = 0;   // 32 bit phase increment, see below
uint32_t ulGrain1PhaseAccumulator = 0; // 32 bit phase accumulator, if we shift >> 20 bits we get a 12 bit value for our output ADC
volatile uint32_t ulGrain1PhaseIncrement = 0;   // 32 bit phase increment, see below
uint32_t ulGrain2PhaseAccumulator = 0; // 32 bit phase accumulator, if we shift >> 20 bits we get a 12 bit value for our output ADC
volatile uint32_t ulGrain2PhaseIncrement = 0;   // 32 bit phase increment, see below

// full waveform = 0 to 4294967296
// Phase Increment for 1 Hz =(4294967296/44100) = 1Hz
// Phase Increment for frequency F = (4294967296/44100)*F
#define SAMPLE_RATE 44100.0
#define TICKS_PER_CYCLE (4294967296.0/SAMPLE_RATE)

// Create a table to hold the phase increments we need to generate midi note frequencies at our 44.1Khz sample rate
#define MIDI_NOTES 128
uint32_t nMidiPhaseIncrement[MIDI_NOTES];

void createNoteTable(float fSampleRate)
{
  for(uint32_t unMidiNote = 0;unMidiNote < MIDI_NOTES;unMidiNote++)
  {
    // Correct calculation for frequency
    Serial.print(unMidiNote);
    Serial.print(" ");
    float fFrequency = ((pow(2.0,(unMidiNote-69.0)/12.0)) * 440.0);
    Serial.print(fFrequency);
    Serial.print(" ");
    nMidiPhaseIncrement[unMidiNote] = fFrequency*TICKS_PER_CYCLE;
    Serial.println(nMidiPhaseIncrement[unMidiNote]);
  }
}

void setup()
{
  Serial.begin(9600);

  createNoteTable(SAMPLE_RATE);
 
  /* turn on the timer clock in the power management controller */
  pmc_set_writeprotect(false);
  pmc_enable_periph_clk(ID_TC4);

  /* we want wavesel 01 with RC */
  TC_Configure(/* clock */TC1,/* channel */1, TC_CMR_WAVE | TC_CMR_WAVSEL_UP_RC | TC_CMR_TCCLKS_TIMER_CLOCK2);
  TC_SetRC(TC1, 1, 238); // sets <> 44.1 Khz interrupt rate
  TC_Start(TC1, 1);
 
  // enable timer interrupts on the timer
  TC1->TC_CHANNEL[1].TC_IER=TC_IER_CPCS;
  TC1->TC_CHANNEL[1].TC_IDR=~TC_IER_CPCS;
 
  /* Enable the interrupt in the nested vector interrupt controller */
  /* TC4_IRQn where 4 is the timer number * timer channels (3) + the channel number (=(1*3)+1) for timer1 channel1 */
  NVIC_EnableIRQ(TC4_IRQn);

  // this is a cheat - enable the DAC
  analogWrite(DAC0,0);
}

void loop()
{
  // read analog input 0 drop the range from 0-1024 to 0-127 with a right shift 3 places,
  // then look up the phaseIncrement required to generate the note in our nMidiPhaseIncrement table
  uint32_t ulInput = analogRead(0);
  ulPhaseIncrement = nMidiPhaseIncrement[ulInput>>3];
 
  // set the phase increment for grains 1 and 2, we do not want a precise frequency for these,
  // they set the character of the note rather than pitch
  ulGrain1PhaseIncrement = analogRead(1)<<18;
  ulGrain2PhaseIncrement = analogRead(2)<<18;
}

void TC4_Handler()
{
  // We need to get the status to clear it and allow the interrupt to fire again
  TC_GetStatus(TC1, 1);
 
  ulPhaseAccumulator += ulPhaseIncrement;   // 32 bit phase increment, see below
  // if the phase accumulator over flows - we have been through one cycle at the current pitch,
  // now we need to reset the grains ready for our next cycle
  if(ulPhaseAccumulator < ulPhaseIncrement)
  {
   ulGrain1PhaseAccumulator = 0;
   ulGrain2PhaseAccumulator = 0;
  }
  else
  {
   // increment the grains
   ulGrain1PhaseAccumulator += ulGrain1PhaseIncrement;
   ulGrain2PhaseAccumulator += ulGrain2PhaseIncrement;
  }
 
  // mix the grains by adding them together - to the result of adding these two 32bit values into the 12 bit ADC Output
  // we shift both values right 21 places to sum two 11 bit values giving a 12 bit result.
  uint32_t ulOutput = (ulGrain1PhaseAccumulator>>21)+(ulGrain2PhaseAccumulator>>21);
 
  // we cheated and user analogWrite to enable the dac, but here we want to be fast so
  // write directly 
  dacc_write_conversion_data(DACC_INTERFACE, ulOutput);
}

Thursday, November 15, 2012

Auduino With Delay

This is the first in a series of posts introducing simple audio effects that can be used in micro controller projects.

Next Week - Bit Crushing effects

The Delay effect is one of the simplest and most effective enhancements we can add to our audio projects.

The delay effect works by recording the output as it is being generated and then mixing this sound back in with itself - after a delay. The result should be familiar to anyone who has every played an electric guitar through an amp with reverb.

In the case of the Auduino synthesizer the result is a mild echo effect and slightly smoother, more metallic sound - the effect can be turned on or off through a push button in the code provided below.

How do we create the delay effect
Delay is very simple to add in a microcontroller, all we need is a block of memory to record the output in.

The larger the block of memory, the longer the delay we can record and the deeper the effect.

In this case we are using a 1K block of memory in the array named sDelayBuffer -

// Duane B
// rcarduino.blogspot.com
// 15/11/2012
// Very simple ring buffer delay
// we record the output in this array
// and then mix it back in with the output as the buffer wraps around
// can be switched on and off by a button on DELAY_BUTTON
#define MAX_DELAY 1024
unsigned char sDelayBuffer[MAX_DELAY];
unsigned int nDelayCounter = 0;
unsigned char bDelay;

The other modification is inside the interrupt service routine which generates the Audiuno output, essentially what we are doing is adding the sound we recorded 1/8th of a second ago on top of the current output value -


  // Duane B
  // rcarduino.blogspot.com
  // 15/11/2012
  // add a button to set bDelay true or false to turn delay on and off
  if(bDelay)
  {
    // Output to PWM (this is faster than using analogWrite) 
    // Here we add the delay buffer to the output value, this produces
    // an subtle echo effect, the delay buffer is effectivley replaying the sound from
    // 1/8th of a second ago.
 
    LED_PORT |= 1 << LED_BIT; // Faster than using digitalWrite
    PWM_VALUE = (output + (sDelayBuffer[nDelayCounter]))>>1;
   
    // add the new output to the buffer so we can use it when the buffer next wraps around
    sDelayBuffer[nDelayCounter] = PWM_VALUE;
    nDelayCounter++;
    if(nDelayCounter == MAX_DELAY)
    {
      nDelayCounter = 0;
    }
  }
  else
  {
    LED_PORT &= ~(1 << LED_BIT); // Faster than using digitalWrite
   
    PWM_VALUE = output;
  }

We test whether delay is enabled, if it is we calculate the output value by adding the initial output to the earlier recorded output from our delay buffer. After outputting this combined value we record it in the delay buffer replacing the value we just used. Over time, the code cycles through the delay buffer over and over again, mixing the current output with a sample from 1/8th of a second back - a bit like playing your instrument in a large hall where the distinct sound is the result of the current sound being constantly mixed with its echo.

Thats all there is to generating delay in a micro controller synth engine - exactly the same code is used to create the delay effect in the RCArduino Five Dollar Synthesizer.


The RCArduino Five Dollar Synthesizer is another audio project enhanced with this delay effect -

http://rcarduino.blogspot.com/2012/10/five-dollar-synthesiser.html 


Further Development
The amount of delay we can provide is determined to the memory we use to record the samples. In the Auduino we are using 1K which at an 8K play back rate gives us 125ms of delay. This can be increased by bit crushing the samples - using 4 bits per sample we get 250ms, 2 bits gets us half a second, with 1 bit we can get a whole second. Unfortunately initial experiments suggest that the effect is largely lost when applying these techniques, its a bit like shouting into a cave and getting a different echo back - your ears just don't buy it.

Auduino Accreditation
The Auduino is an original work by Peter Knight, the original project can be found here -
http://code.google.com/p/tinkerit/wiki/Auduino

Auduino with delay

Auduino with delay is a very slight modification by Duane B (rcarduino) to the original work of Peter Knight.

Notes
- This code also include the volatile fix which allows the Auduino to work correctly in Arduino 1.0 and later
- Remember to use a pull up or pull down resistor if you are not using a push button or switch for the delay button or if your more comfortable modifying the code, replace the button code with true or false.
- LED 13, now indicates whether delay is on or off.


// Auduino, the Lo-Fi granular synthesiser
//
// by Peter Knight, Tinker.it http://tinker.it
//
// Help:      http://code.google.com/p/tinkerit/wiki/Auduino
// More help: http://groups.google.com/group/auduino
//
// Analog in 0: Grain 1 pitch
// Analog in 1: Grain 2 decay
// Analog in 2: Grain 1 decay
// Analog in 3: Grain 2 pitch
// Analog in 4: Grain repetition frequency
//
// Digital 3: Audio out (Digital 11 on ATmega8)
//
// Changelog:
// 19 Nov 2008: Added support for ATmega8 boards
// 21 Mar 2009: Added support for ATmega328 boards
// 7 Apr 2009: Fixed interrupt vector for ATmega328 boards
// 8 Apr 2009: Added support for ATmega1280 boards (Arduino Mega)

#include <avr/io.h>
#include <avr/interrupt.h>

uint16_t syncPhaseAcc;
volatile uint16_t syncPhaseInc;
uint16_t grainPhaseAcc;
volatile uint16_t grainPhaseInc;
uint16_t grainAmp;
volatile uint8_t grainDecay;
uint16_t grain2PhaseAcc;
volatile uint16_t grain2PhaseInc;
uint16_t grain2Amp;
volatile uint8_t grain2Decay;

// Map Analogue channels
#define SYNC_CONTROL         (4)
#define GRAIN_FREQ_CONTROL   (0)
#define GRAIN_DECAY_CONTROL  (2)
#define GRAIN2_FREQ_CONTROL  (3)
#define GRAIN2_DECAY_CONTROL (1)

// DB
#define SMOOTH_PIN 8


// Changing these will also requires rewriting audioOn()

#if defined(__AVR_ATmega8__)
//
// On old ATmega8 boards.
//    Output is on pin 11
//
#define LED_PIN       13
#define LED_PORT      PORTB
#define LED_BIT       5
#define PWM_PIN       11
#define PWM_VALUE     OCR2
#define PWM_INTERRUPT TIMER2_OVF_vect



#elif defined(__AVR_ATmega1280__)
//
// On the Arduino Mega
//    Output is on pin 3
//
#define LED_PIN       13
#define LED_PORT      PORTB
#define LED_BIT       7
#define PWM_PIN       3
#define PWM_VALUE     OCR3C
#define PWM_INTERRUPT TIMER3_OVF_vect
#else
//
// For modern ATmega168 and ATmega328 boards
//    Output is on pin 3
//
#define PWM_PIN       3
#define PWM_VALUE     OCR2B
#define LED_PIN       13
#define LED_PORT      PORTB
#define LED_BIT       5
#define PWM_INTERRUPT TIMER2_OVF_vect
#endif

// Duane B
// rcarduino.blogspot.com
// 15/11/2012
// Very simple ring buffer delay
// we record the output in this array
// and then mix it back in with the output as the buffer wraps around
// can be switched on and off by a button on DELAY_BUTTON
#define MAX_DELAY 1024
unsigned char sDelayBuffer[MAX_DELAY];
unsigned int nDelayCounter = 0;
unsigned char bDelay;

#define DELAY_BUTTON 4


// Smooth logarithmic mapping
//
uint16_t antilogTable[] = {
  64830,64132,63441,62757,62081,61413,60751,60097,59449,58809,58176,57549,56929,56316,55709,55109,
  54515,53928,53347,52773,52204,51642,51085,50535,49991,49452,48920,48393,47871,47356,46846,46341,
  45842,45348,44859,44376,43898,43425,42958,42495,42037,41584,41136,40693,40255,39821,39392,38968,
  38548,38133,37722,37316,36914,36516,36123,35734,35349,34968,34591,34219,33850,33486,33125,32768
};
uint16_t mapPhaseInc(uint16_t input) {
  return (antilogTable[input & 0x3f]) >> (input >> 6);
}

// Stepped chromatic mapping
//
uint16_t midiTable[] = {
  17,18,19,20,22,23,24,26,27,29,31,32,34,36,38,41,43,46,48,51,54,58,61,65,69,73,
  77,82,86,92,97,103,109,115,122,129,137,145,154,163,173,183,194,206,218,231,
  244,259,274,291,308,326,346,366,388,411,435,461,489,518,549,581,616,652,691,
  732,776,822,871,923,978,1036,1097,1163,1232,1305,1383,1465,1552,1644,1742,
  1845,1955,2071,2195,2325,2463,2610,2765,2930,3104,3288,3484,3691,3910,4143,
  4389,4650,4927,5220,5530,5859,6207,6577,6968,7382,7821,8286,8779,9301,9854,
  10440,11060,11718,12415,13153,13935,14764,15642,16572,17557,18601,19708,20879,
  22121,23436,24830,26306
};
uint16_t mapMidi(uint16_t input) {
  return (midiTable[(1023-input) >> 3]);
}

// Stepped Pentatonic mapping
//
uint16_t pentatonicTable[54] = {
  0,19,22,26,29,32,38,43,51,58,65,77,86,103,115,129,154,173,206,231,259,308,346,
  411,461,518,616,691,822,923,1036,1232,1383,1644,1845,2071,2463,2765,3288,
  3691,4143,4927,5530,6577,7382,8286,9854,11060,13153,14764,16572,19708,22121,26306
};

uint16_t mapPentatonic(uint16_t input) {
  uint8_t value = (1023-input) / (1024/53);
  return (pentatonicTable[value]);
}


void audioOn() {
#if defined(__AVR_ATmega8__)
  // ATmega8 has different registers
  TCCR2 = _BV(WGM20) | _BV(COM21) | _BV(CS20);
  TIMSK = _BV(TOIE2);
#elif defined(__AVR_ATmega1280__)
  TCCR3A = _BV(COM3C1) | _BV(WGM30);
  TCCR3B = _BV(CS30);
  TIMSK3 = _BV(TOIE3);
#else
  // Set up PWM to 31.25kHz, phase accurate
  TCCR2A = _BV(COM2B1) | _BV(WGM20);
  TCCR2B = _BV(CS20);
  TIMSK2 = _BV(TOIE2);
#endif
}


void setup() {
  pinMode(PWM_PIN,OUTPUT);
  audioOn();
  pinMode(LED_PIN,OUTPUT);

  pinMode(DELAY_BUTTON,INPUT);
 
  // set pin mode and turn on pull up so that default mode
  // is PENTATONIC, pull the pin low to switch to smooth
  pinMode(SMOOTH_PIN,INPUT);
  digitalWrite(SMOOTH_PIN,HIGH);
}

void loop() {
  // The loop is pretty simple - it just updates the parameters for the oscillators.
  //
  // Avoid using any functions that make extensive use of interrupts, or turn interrupts off.
  // They will cause clicks and poops in the audio.
 
  // defaults to pentatonic stepped tones, pull pin low for smooth frequency without distinct tones
  //    syncPhaseInc = mapPhaseInc(analogRead(SYNC_CONTROL)) / 4;

  syncPhaseInc = mapPentatonic(analogRead(SYNC_CONTROL));
 
  // updated 29/01/2013
  // pull the DELAY_BUTTON pin high for delay, low for no delay
  // use either a pull up/pull down resistor
  // or a pull up resistor with a toggle switch between the pin and ground
  bDelay = digitalRead(DELAY_BUTTON);
 
  // Stepped mapping to MIDI notes: C, Db, D, Eb, E, F...
  //syncPhaseInc = mapMidi(analogRead(SYNC_CONTROL));
 
  // Stepped pentatonic mapping: D, E, G, A, B
 

  grainPhaseInc  = mapPhaseInc(analogRead(GRAIN_FREQ_CONTROL)) / 2;
  grainDecay     = analogRead(GRAIN_DECAY_CONTROL) / 8;
  grain2PhaseInc = mapPhaseInc(analogRead(GRAIN2_FREQ_CONTROL)) / 2;
  grain2Decay    = analogRead(GRAIN2_DECAY_CONTROL) / 4;
}


SIGNAL(PWM_INTERRUPT)
{
  uint8_t value;
  uint16_t output;

  syncPhaseAcc += syncPhaseInc;
  if (syncPhaseAcc < syncPhaseInc) {
    // Time to start the next grain
    grainPhaseAcc = 0;
    grainAmp = 0x7fff;
    grain2PhaseAcc = 0;
    grain2Amp = 0x7fff;
//    LED_PORT ^= 1 << LED_BIT; // Faster than using digitalWrite
  }
 
  // Increment the phase of the grain oscillators
  grainPhaseAcc += grainPhaseInc;
  grain2PhaseAcc += grain2PhaseInc;

  // Convert phase into a triangle wave
  value = (grainPhaseAcc >> 7) & 0xff;
  if (grainPhaseAcc & 0x8000) value = ~value;
  // Multiply by current grain amplitude to get sample
  output = value * (grainAmp >> 8);

  // Repeat for second grain
  value = (grain2PhaseAcc >> 7) & 0xff;
  if (grain2PhaseAcc & 0x8000) value = ~value;
  output += value * (grain2Amp >> 8);

  // Make the grain amplitudes decay by a factor every sample (exponential decay)
  grainAmp -= (grainAmp >> 8) * grainDecay;
  grain2Amp -= (grain2Amp >> 8) * grain2Decay;

  // Scale output to the available range, clipping if necessary
  output >>= 9;
  if (output > 255) output = 255;

  // Duane B
  // rcarduino.blogspot.com
  // 15/11/2012
  // add a button to set bDelay true or false to turn delay on and off
  if(bDelay)
  {
    // Output to PWM (this is faster than using analogWrite) 
    // Here we add the delay buffer to the output value, this produces
    // an subtle echo effect, the delay buffer is effectivley replaying the sound from
    // 1/8th of a second ago.
 
    LED_PORT |= 1 << LED_BIT; // Faster than using digitalWrite
    PWM_VALUE = (output + (sDelayBuffer[nDelayCounter]))>>1;
   
    // add the new output to the buffer so we can use it when the buffer next wraps around
    sDelayBuffer[nDelayCounter] = PWM_VALUE;
    nDelayCounter++;
    if(nDelayCounter == MAX_DELAY)
    {
      nDelayCounter = 0;
    }
  }
  else
  {
    LED_PORT &= ~(1 << LED_BIT); // Faster than using digitalWrite
   
    PWM_VALUE = output;
  }
}


Saturday, October 13, 2012

Algorithmic Music Visualizer

This is a short follow up to a previous post 'Algorithmic Music On Arduino'

http://rcarduino.blogspot.com/2012/09/algorithmic-music-on-arduino.html

The music generation algorithms take advantage of the repeating structure within binary numbers, an obvious development is a visualizer that would reveal some of the structure within the music.

This quick project uses the value of the algorithm output to drive the sound and visuals directly. The visualizer is simply a scrolling buffer of the 8 most recent output values.


While it would be possible to use FFT to visualize the music, in the case of algorithmic music it is more interesting to use the visualizer as a means to explore what is happening within the algorithm rather than in the frequency spectrum.



The short video shows that even with no additional processing, the visuals are very different for each of the algorithms and reveal some of the structure within the music.

The first algorithm running from 0:11 to 0:38 produces a cascade effect reminiscent of the matrix - unfortunately this effect is subtle and needs development for the best effect. The second section of the video provides a more obvious demonstration of how the sound is being developed within the algorithm.

Some other ideas to explore in the future -

1) Bit rate - for each bit in the 8 bit output, increment a counter if it has changed from the last sample. Use the values for the 8-bits to drive the display

2) Divide by 2 - Use the current output for row 0, update row 1 every second output, row 2 every fourth output, row 3 every 8 etc upto 128

For now, here is a scrolling buffer of the 8 most recent output values (and a little reaction from the family)


Duane B

Saturday, September 22, 2012

Algorithmic Music On Arduino

Original By Viznut Arduino Version

This is an interesting little project that can be built in minutes. It plays 13 different tunes all of which are generated from a single line of C code, the tunes are selected through four push buttons which have 16 possible combinations, including silence and two free slots for your own music.

The project also includes an 8 LED visualizer that is driven by the same line of code as the music.

About Algorithmic Music


The original tunes were produced by Viznut and by others in response to his original blog post here -

http://countercomplex.blogspot.com/2011/10/algorithmic-symphonies-from-one-line-of.html

The original Arduino port was completed by Arduino Forum user Stimmer and can be found in this post -

http://arduino.cc/forum/index.php/topic,74123.msg558213.html#msg558213

Arduino Forum user Zeni also has an interesting variation which allows the user to enter new algorithms at runtime, this can be seen in action in the forum topic above.

The RCArduino Version adds the ability to select the current algorithm and the 8 LED Visualiser.

The RCArduino Version code and schematic can be found below, if your interested in the theory behind the music or demoscene in general ( http://en.wikipedia.org/wiki/Demoscene ) you can pick up the trail from Viznuts original video linked at the start of this post.

An application in Grain Synthesis ?
 
One area that would be interesting to apply these techniques is grain synthesis. Grain synthesis is based on repeating a simple grain of sound while adding additional grains, envelopes, oscillators or filters to make the sound more interesting. These operations can be computationally expensive and do not always produce interesting sound. The techniques used to generate the algorithmic music from a simple counter demonstrated by Viznut could also be applied to generating more interesting synthesis grains.

The best known example of an Arduino Grain Synthesizer is the Auduino, its incredible to see that all of the sound generated by an auduino is the result of overlaying just two triangle waveforms. In one of the videos linked below you will see the triangle waveforms interacting together with the rich sound that results.

Examples of built Auduino Synths can be found at the end of this post - 
http://rcarduino.blogspot.com/2012/08/adding-audio-to-arduino-projects.html

An interesting development of the grain synthesis would be to combine it with the realtime code upload work of Zeni, it should be possible to create grains that modulate and transform themselves to create a more interesting sound palette.

RCArduino Version - Based on original work of Viznut and original Arduino port of stimmer



// one-line algorithmic music
// see viznut's blog http://countercomplex.blogspot.com/2011/10/algorithmic-symphonies-from-one-line-of.html
// and http://www.youtube.com/watch?v=GtQdIYUtAHg&feature=related

// ported to arduino by stimmer
// Audio out on pin 10

// Further ported to Interrupt based music generation with 4 button selection of upto 16 algorithms (or 15 + silence on no buttons pressed)
// by Duane B aka RCArduino
//
// Buttons are digital 8,9,10,11, music output is on digital 6
//
// Also added a four LED Visualiser on digital pins 2,3,4,5
//
// Update - LED Visalizer is now 8 bit using digital pins 2,3,4,5 and analog pins 0,1,2,3


#define SAMPLE_MAX (65535.0)
#define SAMPLE_FREQUENCY (8000.0)
#define TIMER1_FREQUENCY 2000000
#define UPDATE_RATE 8000

// by keeping t global we can use it to drive the visualiser as well
long t;

// iterate the grains and LFO
SIGNAL (TIMER1_COMPA_vect)
{
 OCR1A += (TIMER1_FREQUENCY/UPDATE_RATE);

 t++;

 switch(PINB&15)
 {
   default:
   case 0:
     OCR0A = 0;
     break;
   case 1:
     OCR0A = ((-t&4095)*(255&t*(t&t>>13))>>12)+(127&t*(234&t>>8&t>>3)>>(3&t>>14)); // by tejeez
     break;
   case 2:
     OCR0A = t*(t>>11&t>>8&123&t>>3); // by tejeez
     break;
   case 3:
     OCR0A = t*((t>>9|t>>13)&25&t>>6);   // by visy
     break;
   case 4:
     OCR0A = (t*(t>>5|t>>8))>>(t>>16);   // by tejeez

     break;
   case 5:
     OCR0A = ((t*(t>>8|t>>9)&46&t>>8))^(t&t>>13|t>>6); // by xpansive
     break;
   case 6:
     OCR0A = ((t&4096)?((t*(t^t%255)|(t>>4))>>1):(t>>3)|((t&8192)?t<<2:t)); // by skurk (raer's version)
     break;
   case 7:
     OCR0A = (t>>7|t|t>>6)*10+4*(t&t>>13|t>>6); // by viznut, xpansive, varjohukka
     break;
   case 8:
     OCR0A = t*5&(t>>7)|t*3&(t*4>>10); // by miiro

     break;
   case 9:
     OCR0A = (t|(t>>9|t>>7))*t&(t>>11|t>>9); // by red
     break;
   case 10:
     long v;
     OCR0A = v=(v>>1)+(v>>4)+t*(((t>>16)|(t>>6))&(69&(t>>9))); // by pyryp
     break;
   case 11:
     OCR0A = (t>>6|t|t>>(t>>16))*10+((t>>11)&7);  //by viznut
     break;
   case 12:
     OCR0A = (t*(4|7&t>>13)>>((~t>>11)&1)&128) + ((t)*(t>>11&t>>13)*((~t>>9)&3)&127); // by stimmer

     break;
   case 13:
     // free to use
     break;
   case 14:
     // free to use
     break;
   case 15:
     // free to use
     break;
     // any more and we need another bit from PORTB
 }
}

void setup()
{
  TCCR1A=0x0;          // set the timer prescaler to 8 = 16/8 = 2MHz
  TCCR1B=0x02;          // set the timer prescaler to 8 = 16/8 = 2MHz
  TIMSK1 |= (1<<OCIE1A);   // Enable output compare match interrupt on OCR1A
  
  //TCCR0A=0B10110011;                                    //-8 bit audio PWM
  TCCR0A=0B10000011;                                    //-8 bit audio PWM
  //TCCR0A=0x83;          // Set timer waveform generation mode to FAST PWM, clear OC0A On match, set at bottom - OC0A = digital pin 6.
  TCCR0B=0x01;          // Set to clock frequency, no prescaler
  OCR0A=127;            // set in the middle - do we need this ? probably not.
  DDRD|=1<<6;          // Set digital pin 6 to output - channels 2 and 3

  DDRB &= (~15); // set digital pins 8,9,10,11 as inputs
  DDRD |= ((1<<2) | (1<<3) | (1<<4) | (1<<5)); // set digital pins 2,3,4,5 as outputs for the visualiser
  DDRC &= (15);
}

void loop()
{
 unsigned char output = OCR0A;

 // clear visualiser bits on portD
 PORTD &= 0B000011;

 // set the portd part of the visualiser using the top 4bits of output
 PORTD |= ((output>>4)<<2);

 // set the portc part of the visualiser using the bottom 4 bits of output
 PORTC &= 0B110000;
 PORTC |= (output & B1111);
}




For a simple amplifier circuit to use with Arduino projects see here -

http://rcarduino.blogspot.com/2012/08/adding-audio-to-arduino-projects.html

I am currently working on an Arduino based modular synthesizer with a very different sound, stay tuned

Duane B.

Sunday, August 26, 2012

The Must Build Arduino Project - The Illutron B

Is there a must build Arduino project ? Something simple that can be built in minutes and tinkered with for days or weeks ?

There is now.

Presenting the Illutron B -




The Illutron B is a development of the Illutron synthesizer originally created by Nikolaj Mobius. All of the sound in the clip is being generated by the Arduino with no outside assistance or post processing - the bass notes are extraordinary.

Even more incredibly is that all of the sound is being generated using just one analog output.

How does it work ?
If you think of an analog output as a power switch, the longer it is switched on, the more power we output. By varying the duration that the power is on over time we can output a waveform, for example a sine wave.

Its a technique known as Direct Digital Synthesis. The is a good example here http://interface.khm.de/index.php/lab/experiments/arduino-dds-sinewave-generator/


Taking it further - Skip the theory if you like, or read on -
The Illutron B is a wave table synthesizer, this is a synthesizer which uses the Direct Digital Synthesis described above to generate sound waves using descriptions 'wave tables' that are stored in the memory. 

The wavetables are simply arrays, if you take the values from each of the wave header files and copy them into your favorite spreadsheet you will see the following - 


Synth designers know that as listeners we do not find endlessly repeating sine, square or triangle waves very interesting. To make the sound more musical wave table synthesizers combine the waveform with an envelope. 


The envelope describes the power or loudness of the waveform over time - think of a snare drum, its starts very powerfully and decays away to nothing very quickly - a flute can start softly, reach a level then decay softly - its the envelope that describes this. The simple trick of combining the wavetable with an envelope is responsible for the huge range of musical possibilities in wavetable synthesis.

How do we do this in software ? - It couldn't be easier, we mulitply the waveform by the envelope, as the envelope value gets smaller, so the output waveform shrinks away to nothing.

Have a look through the code, its extensively documented.

Built in envelopes - the faster the sound drops away, the more percussive (drum like) the envelope -


You can add your own envelopes, or modify the supplies ones, there is no rule that an envelop cannot start quietly and get louder - or look like a heart beat for a pulsing bass sound.


Build your own pocket night club -

How long does it take to build ? anywhere from 5 to 10 minutes, seriously, if you have two potentiometers a few capacitors and resistors you can be playing your own Illutron B in 10 minutes.

What does it do ?

While the synth is very powerful, Nikolaj has included a demonstration tune which is the basis of the clip.

In the clip I am using the Illutron B which provides control over the beats per minute and also the pitch of one of the four channels using just two potentiometers to jam with the demo tune.

This is where RCArduino comes in. When I first heard the Illutron I was blown away, it is far and away the best Arduino Audio project of all. The drum sounds are awesome, the bass is incredible, you will not get tired of exploring those low notes. I know that as a community we can make great things from this.

For this reason I have spent the past few days refactoring, optimizing and documenting the Illutron version B.

Some ideas for your own Illutron projects -

1) Hook up some peizo knock sensors for an electronic drum kit
2) Bass loop generator with push buttons for different bass drops
3) A four channel sequencer
4) Sound effects for games, installations and robots
5) Its a great basis for a Thermin or similar physical instrument.
6) Add some soft pots, cross faders or a stylus to control the pitch of the different channels

Nightclub in your pocket

My own idea is to build on the current example - Ideally I would like to see a set of beats or tunes included in a 'Night club in a box' where the user can select the beat with a push button. The user is then free to jam with the selection in the same manner as the video - Arduino, two potentiometers, one resistor, two capacitors, synched LED Light show and you have an instant pocket night club that everyone can enjoy.


In order to support the further development of Nikolaj's original concept I have refactored the original Illutron code into a more user friendly class library while at the same time taking the opportunity to do some optimizing and improve the readability of the code.

If you would prefer to use Nikolaj's original code for the Illutron Wave Table Synth, its available here - http://www.instructables.com/id/Turn-your-Arduino-into-a-4-voice-wavetable-synth-w/

The Illutron B code is being added to the Illutron repository on github I will provide a link once its uploaded,  in the meantime, you can get it from me - DuaneB on the Arduino forum.

If your a musician it should be immediately obvious how to setup and trigger the different voices, the sample sequencer is also easy to understand and modify. If like me you are not a musician, build one anyway, the demo tune is great to play with and I hope to get some new tunes included for you to upload.

Whats an Illutron ?

Its actually an art studio on a ship in Copenhagen harbour. The original Illutron synth was created by a member of the Illutron collective Nikolaj Mobius. You can find out more about Illutron and their work in light, sound, electronics and a surprising amount of fire here - http://illutron.dk/posts/54


Building your own Illutron B - 

The easy part - 
2 * 10 K Potentiometers
4 * LEDs (or eight if you like)
4 * LED Current Limiting Resistors (500 to 1000 Ohms should be fine)

The only slightly less easy part -

Jelly Beans

What are Jelly Beans and why do I need them ?

Jelly Beans are those common components that all circuits need and you should have a jar full of. If your just starting out, you might not have these, but get some, they are very cheap standard components which are widely used in all sort of circuits.

1 * 2.2K resistor (or two 1K resistors in series - anything thats close)
1 * 0.1uF capacitor - you should have hundreds of these, they are used for stopping interference from other components from reaching the sensitive parts of your circuits, if you don't have any, buy 20, they are cheap and you will use them for everything.
1 * 100uf  capacitor - again these are widely used. In this case the capacitor is there to filter out a DC voltage so that your Audio equipment only receives the alternating signal part of the Arduino output. If you don't have any of these, get 10.

Here is the Schematic -

Note : The Fritzing software used to draw the circuit labels C2 with 0.1mF, this is the 100uf capacitor, 100uf is the most common way to represent this value but 0.1mf is also correct (1uf = 0.001mf so 100uf = 0.1mf)

You can also follow the original build instructions here -http://www.instructables.com/id/Turn-your-Arduino-into-a-4-voice-wavetable-synth-w/

Here is mine as used in the video. For the nightclub in a box project I will be adding two amplifiers based on the previous rcarduino post - Adding Audio To Arduino Projects - http://rcarduino.blogspot.com/2012/08/adding-audio-to-arduino-projects.html


The final part of the build to to plug the Illutron B into your MP3 Player docking station - I used a section of head phone cable soldered to a three pin header for this -


The code is now available for download on the RCArduino downloads page here -

http://rcarduino.blogspot.ae/2014/02/rcarduino-downloads.html

If you have any problems with the download, contact me 'DuaneB' through the Arduino forum or leave a comment.

In a future post I will explore more of the capabilities of the Illutron B, the Illutron team will also be working on new Illutron synth based projects.

Stay tuned

Duane B