Beginner Digital Synth [closed]

Answer 1

1. This problem is basically about mapping functions to arrays of numbers. A language that supports first-class functions would come in really handy here.

2. Check out http://www.harmony-central.com/Computer/Programming and http://www.developer.com/java/other/article.php/3071021 for some Java-related info.

3. If you don't know the basic concepts of encoding sound data, then read http://en.wikipedia.org/wiki/Sampling_rate

4. The canonical WAVE format is very simple, see http://www.lightlink.com/tjweber/StripWav/Canon.html. A header (first 44 bytes) + the wave-data. You don't need any library to implement that.

In C/C++, the corresponding data structure would look something like this:

typedef struct _WAVstruct
{
    char headertag[4];
    unsigned int remnantlength;
    char fileid[4];

    char fmtchunktag[4];
    unsigned int fmtlength;
    unsigned short fmttag;
    unsigned short channels;
    unsigned int samplerate;
    unsigned int bypse;
    unsigned short ba;
    unsigned short bipsa;

    char datatag[4];
    unsigned int datalength;

    void* data; //<--- that's where the raw sound-data goes
}* WAVstruct;

I'm not sure about Java. I guess you'll have to substitute "struct" with "class" and "void* data" with "char[] data" or "short[] data" or "int[] data", corresponding to the number of bits per sample, as defined in the field bipsa.

To fill it with data, you would use something like that in C/C++:

int data2WAVstruct(unsigned short channels, unsigned short bipsa, unsigned int samplerate, unsigned int datalength, void* data, WAVstruct result)
{
    result->headertag[0] = 'R';
    result->headertag[1] = 'I';
    result->headertag[2] = 'F';
    result->headertag[3] = 'F';
    result->remnantlength = 44 + datalength - 8;
    result->fileid[0] = 'W';
    result->fileid[1] = 'A';
    result->fileid[2] = 'V';
    result->fileid[3] = 'E';

    result->fmtchunktag[0] = 'f';
    result->fmtchunktag[1] = 'm'; 
    result->fmtchunktag[2] = 't';
    result->fmtchunktag[3] = ' ';
    result->fmtlength = 0x00000010;
    result->fmttag = 1;
    result->channels = channels;
    result->samplerate = samplerate;
    result->bipsa = bipsa;
    result->ba = channels*bipsa / 8;
    result->bypse = samplerate*result->ba;

    result->datatag[0] = 'd';
    result->datatag[1] = 'a';
    result->datatag[2] = 't';
    result->datatag[3] = 'a';
    result->datalength = datalength;

    result->data = data; // <--- that's were the data comes in

    return 0; // an error code, not implemented, yet ...; in Java: return result
}

Again, I'm not sure about Java but the conversion should be straightforward if you convert the void-pointer to an array corresponding to the bitrate.

Then simply write the entire structure to a file to get a playable wave file.

Answer 2

While studying for my degree, my dissertation project was the creation of a Java based modular synthesizer, and the University at which I studied saw fit to make my report publicly available:

A Software Based Modular Synthesiser in Java

Answer 3

I dont't know if that helps, but if you can use MIDI for anything, you should check out JFuge.

Answer 4

Check out Frinika. It's a full-featured music workstation implemented in Java (open source). Using the API, you can run midi events through the synthesizer, read the raw sound output, and write it to a WAV file (see source code link below).

Additional information:

• Frinika Developer Area
• Source code for midi renderer tool