@whale-av oh and another thing, there would have to be a lot more sine waves, because morphing timbre cannot occur with one sine wave, otherwise you´re just manipulating that sine waves´s volume, what I mean is that there are 16 sine waves or oscillators PER curve, so the morphing occurs by manipulating the volumes of the set of 16 sine waves per curve,. Lets have mercy on my computer and for now use 5 sine waves as a set for morphing timbre per curve. is it going to take to put a set of 5 graphs and oscillators per curve? (there are 31 curves for timbre), i wish but i think it cannot be done, to use the five graphs for all the curves? that´s impossible right? would there need to be 155 graphs and oscillators? because the set of 5 sine waves is there just to make and fuse into ONE sound, the one that will go to one curve and morph by manipulating it´s volumes.

dbE2.wav dbE1.wav A2.wav C2.wav A1.wav G2.wav D2bass.wav C3pizz2.wav B1pizz.wav E2pizz.wav Fsharppizz.wav

Hi David, Thanks so much. I've uploaded the sound files, as they're quite small. It'll make it easier as they'll just loadbang into the tables if you have them in the same folder. I really appreciate your looking at that. I'll have a look at those links carefully. -George

@whale-av Hello again, thank you very much for replying so quickly!

Ok so, the computer will always be a PC and will always have Pd installed so that makes things a lot easier I suppose (not sure what version it will be though, but since all those computers have only recently been equipped with Pd I'm assuming it's either vanilla or extended so it should be ok).
I just put my 17 sound files in the same folder and named them 1.wav ...... 17.wav. They are in the same folder as your patch now too.
Well, the numbers are not exactly "set", in fact, each sound file corresponds to a certain range of numbers. Ugh, I did it again, forgot to mention the range... Sorry..... I'd probably need a different object for a range... It's still kinda difficult for me to think in "computer" language about things related to music so that's why I'm forgetting to mention tons of stuff, but I'll try to fix that in this reply, sorry again!
I will just post the table to make it easier for you to understand.

But first, I would just like to explain how the table works. First I need to ask if it's possible to attribute the same numbers/range to different sound files, depending on the number's "place" within the combination? (You'll understand once you see the table.) Maybe it isn't and in that case I should think of a way to change the table according to what can be done?
So, there should be 13 numbers in total within the combination (13 numbers to type, like 13 empty slots waiting to be filled with numbers), but I said there are five sub-combinations. The first, second and fourth are 2 digit numbers, the third is a 3 digit number and the fifth is a 4 digit number. Obviously, three of these are 2 digit numbers, but the thing is I can't really choose the ranges (I have a set range for each sub-combination and can only try to figure out the easiest way to work with them) and the ranges are actually intertwined, So some numbers can be repeated, that's why I'm asking if the same numbers can be attributed to different sound files based on whether it's the first, second or fourth sub-combination. If that's too complicated or even impossible, I can only think of sacrificing a couple of sound files and just let the program play the same file twice instead.

Also important: is it possible to "condition" to program into playing a sound file based on the first combination, the first 2 digit number? Simply put through simple chords, say in the first combination you can type either 01 or 02, where 01 is A major and 02 is B major. Next, you can type in either 02 or 03, where 03 is E major (irrelevant right now), but 02 can be either D major or F sharp major (not A major like in the first combination), based on what the previous number was. If it was 01 (A major), then this time 02 will be D major, but if the first number was 02 (B major) then the 02 in the second combination should be F sharp major. IS something like this even possible, or do I need to figure out a way to go around this and alter my table? I CAN alter the table, but I can't work with the ranges of the combinations, or the number of the combinations, it must be 5.
Lastly, can different numbers be attributed to the same sound file based on their position within the combination?

Ok so finally here's the table, and I will just give you my current table so you can tell me if it's possible to do. I understand if it's too complicated and I need to change it! Maybe you can give me some tips on how to change it to make it the easiest to do in Pd.
1st SUB-COMBINATION
01-19 - 1.wav
20-22 - 2.wav
23-31 - 3.wav
2nd SUB-COMB.
01-03 - 4.wav (if the first was 01-19 or 20-22; or shorter, 01-22)
01-03 - 5.wav (if the first was 23-31)
04-06 - 6.wav
07-09 - 7.wav (if the first was 01-19)
07-09 - 8.wav (if the first was 20-31)
10-12 - 9.wav (if the first was 01-19)
10-12 - 10.wav (if the first was 20-22)
10-22 - 11.wav (if the first was 23-31)
3rd SUB-COMB.
900-975 - 4.wav (if the first was 01-19)
900-975 - 2.wav (if the first was 20-22)
900-975 - 12.wav (if the first was 23-31)
976-999 and 000-020 - 1.wav (if the first was 10-22)
976-999 and 000-020 - 13.wav (if the first was 23-31)
4th SUB-COMB.
01-09 - 7.wav (if the first was 01-19)
01-09 - 8.wav (if the first was 20-31)
10-69 - 14.wav (if the first was 01-19)
10-69 - 3.wav (if the first was 20-31)
70-99 - 6.wav (if the first was 01-22)
70-99 - 11.wav (if the first was 23-31)
5th SUB-COMB.
(this one is arbitrary so no matter what the last 4 numbers are, the sound file that is triggered should be based on the first combination, or something like that, so...)
xxxx - 15.wav (if the first was 01-19)
xxxx - 16.wav (if the first was 20-22)
xxxx - 17.wav (if the first was 23-31)

Now, I need to thank you if you had the patience to actually read all that and think about it..... I can only hope you could understand the idea, but please, if this is just too complicated, do tell me to change the table. Maybe, if you can or think I will understand, try to explain how the ranges work when playing sound files and how I can use that to my advantage to still keep all the sound files (or as many as possible) but make the thing easier to program.

Now I know I had a few more questions but I can't remember right now and I don't want to bombard you with everything all at once, I realize even this is way more than I could have asked for, you have been a huge help, really! Even if I don't entirely understand every single element of your patches, I did study them and used the help option like you advised me to, so I really am trying to get the hang of this. I completely understand if you just want to take a break from this if it's too complicated and time-consuming, I just ask that you give me some kind of feedback so I know what to expect!

Anyway, thank you for all your help and guidance until now, and for the guidance you'll provide in the future if you so choose!

because each odd partial cancels itself out with phase cancellation. Halfway through the wave, every odd partial (sine wave) will also be halfway through a cycle. Therefore, since the sine wave would be negative in the second half of its wave, adding it to the positive part from the beginning of the wave cancels it out. This works for different starting phases also, because whatever phase the partial started at, halfway through the wave will still be the opposite polarity
It's helpful to picture it: picture 3 cycles of a sine wave. Cut them in half, and you will see that there are 1 1/2 cycles in each half, and the second half of the overall wave starts on a negative half of a sine wave

Hello everyone!
You have all been very generous in your time and helping me work out the basics of PD.
Please take a look at the game I have developed and any suggest improvments I could make!

(there are 3 wav files that need to be kept in the same folder as the game)

Many thanks,

Peter

mariowins1.wav mariodies1.wav gameover1.wav GAME7.pd

soundfiler: no method for 'Metalbowl1pd.wav'
soundfiler: no method for 'Metalbowl1pd.wav'
soundfiler: no method for 'Metalbowl1pd.wav'
soundfiler: no method for 'Metalbowl1pd.wav'
soundfiler: no method for 'Metalbowl1pd.wav'
soundfiler: no method for 'Metalbowl1pd.wav'
soundfiler: no method for 'Metalbowl1pd.wav'

Im going to bed.

@LiamG tom.wav snare actual.wav pd project daragh.pd kick.wav hat.wav chord.wav

Thank you all for your replies, I didn't expect such feedback. Especially thanks jamesmcn - everything was described well, I just have something to add:

1 and 2. The top section performs two functions - it's smth. like "probability generator" (leftmost part), which, in accordance with LFO, defines how often droplet sound is being generated at the moment, and (rightmost one) an array containing tones, at which [vcf~]s of certain droplets resonate (0,2,3,5,7,8,10 in it mean C-minor scale). And every 5 seconds [route] picks these tones (notes) up from the array randomly and sends them to [vcf~] objects after [stream~] abstractions. These tones are transformed into frequencies in Hz to control bandpass filters.

3. Yes, the [stream~] is the meat of the synthesis, but it is not "very carefully" filtered noise, although should be. I just tried to make it sound like water droplets, and made some variations in cutoff frequency (first creation argument) and stream density (2nd creation argument) to make droplets sound more diverse. The stream~.pd itself is actually a very simple noise generator, and this is the oldest part of the patch itself. I wanted to make something like a rain noise, made this abstraction, and put it off until it came into play.

4 and 5. You know, if you delete the mixer/processor section (except reverb), you may not notice much difference: it just makes left and right channel slightly different from time to time - for merciless freezeverb to mix up left and right channels in one stream anyway. Buy the way - [freezeverb] is just an enhanced version of that you can see in help -> browser -> G08.reverb.pd. The only serious difference is that it uses delay_time_counter.pd, which calculates the times for delay lines in accordance with this formula: t = t1/2^(n/numlines)-t1/2, where t1 = the largest early reflection delay time value, numlines = total delay lines number (28 here), and n = current delay number (starting from 0). I found this algorithm here: http://musicdsp.org/archive.php?classid=4#44 but changed it a bit (actually, added "-t1/2" to make echoes appear earlier. I still don't understand completely how [freezeverb~] works. To be more precise, I don't understand what actually does [early_reflection_delay_line] do - but Miller Puckette in his example applied similar [reverb-echo-del] abstraction, and it works well! It makes a "power-preserving" mix, very useful thing in recirculating reverbs.

6. Two sine wave oscillators take their frequencies from two first randomly picked up notes from an array (see item 1 and 2). There are also two frequency modulators, 1846 Hz and 4 Hz sine waves, to saturate their spectrum. So it sounds a bit like noise, mainly because there are already too much sine waves from the [stream~] abstraction, and I thought it worth adding something at higher frequencies. And reverb smoothes these oversaturated sine waves, making them sound noisy.

7. How reverb similar to [freezeverb] works is described in help browser, I just can't understand why power preserving mix works. Also I tried to make one stereo reverb based on Miller Puckette's model, but a couple of experimental ones failed. This one is my best reverb ever.

8. Yes, and [master] abstraction is just a place where volume control, or spectrum analysis, are handy to perform from. Something to put all the wires at, and to listen to its output.

Hi - here is a partial answer to your first question. I'm just working through this myself so if anyone can add to this, please do. First, if you are not quite sure on the math here is a link -

. And second, the sine and cosine functions are used to generate sine waves. Sine waves can be used for all sorts of things - supposedly any sound, or a lot of sounds, can be created by combining sine waves. The attached patch should demonstrate how a sine wave is created using trig functions and then how different waveforms can be generated using multiples of a single sine wave.
I hope this helps.
Simon

that's a crummy sounding sine wave in my patch, but it should give you the basic idea

http://www.pdpatchrepo.info/hurleur/sine_function_demo.pd

This is a collection of abstractions that generate bandlimited oscillators. They include:

[bl-saw.mmb~] - bandlimited sawtooth waveform
[bl-pulse.mmb~] - bandlimited pulse wave with PWM
[bl-tri.mmb~] - bandlimited triangle wave
[bl-asymtri.mmb~] - bandlimited asymmetrical triangle wave (sort of...see below)

There is also an object called [bl-init.mmb]. This is the object that initializes all the waveforms and at least one instance MUST be included in order for the others to work.

There are also help patches included.

IMPORTANT!
Before you can use these, you must do the following steps.

1. Open [bl-init.mmb]
2. There is a message box that says [44100(. This is the maximum sampling rate that these will work at (running at lower sampling rates will be fine). If you plan on using higher sampling rates, change this message box and click it. Technically, it will still work at a higher sampling rate, but it won't generate harmonics above the sampling rate in this box.
3. Click the [bang( to fill the wave tables. This patch actually creates a wavetable for EVERY harmonic between 30Hz and the Nyquist frequency. So it will take a few minutes. Be patient! You will get a message in the Pd window when it is done.
4. Save the patch.

Once you do this, [bl-init.mmb] will simply load with the tables already generated, so you don't have to wait every time you instantiate it. I didn't have this already done for you in order to keep the upload small, and so you can see how to adjust it if you need to.

So, I guess I'll go ahead and try to explain how these work. As stated above, every harmonic is generated in [bl-init.mmb] for the oscillators. It doesn't create a table for each set of harmonics however (e.g., there isn't a saw table with two harmonics, a saw table with three harmonics, etc.). Instead, each of these individual tables are tacked on to the end of each other to create one long wave table. So, for each set of 1027 samples in the sawtooth wavetable, there is one cycle with a set amount of harmonics.

When the oscillators read the frequency input, it is divided into the Nyquist frequency to determine how many harmonics are needed. It then uses this (* 1027) as the offset for the table. This is how I got around the problem of table switching at block boundaries. By doing this way, the "switching" is done at audio rate.

There are actually two [tabread4~]s. One has one less harmonic than the other. As the frequency changes it crossfades between these tables. When one table goes completely silent, that's when it "switches." Below 30Hz, they switch to geometrically perfect waveforms.

[bl-saw.mmb~] and [bl-tri.mmb~] just read through the tables. Nothing really interesting about them.

[bl-pulse.mmb~] is actually the difference between to sawtooths. In other words, there are two bandlimited sawtooth oscillators inside of it. Adjusting the pulse width cause the phase of one of the sawtooths to shift. When you subtract this phase-shifted sawtooth from the other, it creates a bandlimited pulse wave...without oversampling! This is the same Phase Offset Modulation method used in Reason's SubTractor.

[bl-asymtri.mmb~] uses the same technique as [bl-pulse.mmb~], except it uses bandlimited parabola waves instead of sawtooths. Adjust the phase offset sets where the top vertex is. This doesn't really generate true triangle or saw waves, though. They still have the parabolic curve in them, so the harmonics seem to come out a little more. It's more of a "reasonable approximation." But, it is bandlimited, and it does sound pretty cool to modulate the shape. I don't have the scaling quite right yet, but I'll get to it later...maybe.

I should also mention that these use my [vphasor.mmb~] abstraction, so the phase reset is sample accurate.

I'll eventually set these up to allow frequency and pulse-width arguments, but I'm currently in the process of moving to another country, so it may be a little bit before I get around to it.

Didn't any of that make any sense?

http://www.pdpatchrepo.info/hurleur/bl-mmb.zip