I thought I would be able to figure this common problem by myself but I'm really stumped and becoming too tired to think mathematically.

My problem is that I am trying to sweep the frequency of a bandpass filter up and down. I want to cover the majority of the audible spectrum with this filter sweep, but when it sweeps up and down, I want it to sound like it is sweeping "evenly."

If I just use a linear function to sweep up and down, moving the frequency up and down, it doesn't sound even, because it takes a lot longer to sweep from 10000 hz to 20000 hz than it does to sweep from 100 hz to 200 hz, even though the difference in frequencies in both cases is an octave, which sounds like the same difference in pitch in terms of psychoacoustics.

When the filter sweeps up and down, it has plenty of time to showcase the high frequencies, but as it reaches the lower frequencies it just flies right through them. I'm wondering what sort of equation I can use to change this, so that the time it takes for the filter to sweep through one musical interval is constant, regardless of where in the spectrum it is.

I'm really tired and confused after spending too much time playing with exponents and logarithms and wishing that I had payed more attention in my math classes. I hope I've still been able to make myself clear.

A thought just occurred to me. When you are using speakers, the sound from the left speaker will affect the sound from the right speaker, so if the same signal is present in both channels but it's delayed in one, the phase difference can cause the signals to be added or subtracted.

In the "sweet spot" between the two speakers, if the two channels are out of phase, you will hear certain frequencies being amplified or attenuated.

If I am listening with headphones, I can hear the signals independently in each ear, so even if I'm listening to two sine waves that are 180 degrees out of phase, I will be able to hear both independently, and I will interpret the difference in phase as a delay that will effect my spatial imaging of the sound.

If I were listening with speakers instead, wouldn't this same situation with two sine waves that are 180 degrees out of phase cause the sound to be completely attenuated (depending on where I am seated in the room)? Does this mean that mixing with headphones can lead me to overlook some issues that I would normally hear with speakers?

Sorry, I guess this isn't really a PD-specific question; it's more of a general audio production thing that will be important no matter what applications I'm using.

Hey everyone!

I've been working on this patch a little bit at a time for the past two weeks or so. It's nothing too complicated, but I really like how it has turned out.

Visit my brand spanking new soundcloud page to listen or just download the patch in the attachment and tweak it yourself.

http://www.pdpatchrepo.info/hurleur/add2.zip

Hey everyone, sorry to bother you with such a newbish question, but I'm looking for an easier way to make sequences in PD.

I'm a windows user, and I just downloaded and installed seq24. It seems pretty easy to set up sequences for different instruments using seq24, but how do I get it to send the data to PD? I tried just setting the MIDI data to be sent to channel 1 using the Microsoft MIDI Mapper, and then in PD I set the driver to ASIO so it would use the Microsoft Sound Mapper... I guess this doesn't have anything to do with MIDI though.

I tried just making a [notein 1] object and attaching some number boxes to it, and then I started to play a sequence in seq24. I guess it's not that simple.

Anyway, I probably look like an idiot for not knowing how to set this up, but can someone point me in the right direction anyway?

Hey guys, I'm new here and I'm pretty new to Pure Data. I wasn't sure whether to post this in the output~ section or the patch~ section, but it probably doesn't matter that much.

Here is a fairly small patch that I've been working on for the past day or so. I'm using two 3-oscillator FM synths that are playing random minor notes to form different chords. One synth has a short attack, medium decay, and a noticeable delay, while the other has a long attack and acts as a sort of synth pad.

I also added some drums into the mix, but I'm not really satisfied with them.

Anyway, it's nothing phenomenal and the guts form a very disorganized mess (I probably should have used more subpatches). Still, I like the sound and the mood it's producing... it kind of reminds me of a bowed vibraphone.

Check it out if you get the chance - It should just start if you press the big button but I'm fairly new to all of this stuff so I apologize if I screwed up the patching somewhere and it doesn't work (I think I'm getting a stack overflow error but I'm not sure where it's coming from or what it's doing. It happened when I copied the contents of extractions into subpatches and deleted the abstractions to make it fit in one folder). Feedback would be much appreciated!

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

Oh, I see! I actually like that solution a little bit better, but I don't seem to have an [exp~] as a signal object (I do have the non-signal version). I guess I could just use [expr~] and approximate e instead.

Anyway, I can see this being very useful. Thanks for the help!

I just realized that I can use mtof~ to do what I want by just sending it values between 0 and about 135, and section 3.1.1.4.3 of Johannes Kreidler's PD Tutorial seems to have an explanation of how this works using an exponential function. I guess the answers are all here for me...

Hmmm, I was actually going through the FM tutorials in MSP a few nights ago. There were some great sounds even in that simple patch, but nothing quite like the stuff in this patch. I guess it just shows how much you can get out of a little variation and weighted randomness.

Out of curiosity, does anyone know if this is a legitimate Autechre patch, or is it something someone else made in their style? Either way, it's pretty awesome. I'm surprised that all of the sounds seem to be synthesized through FM; I have no idea how to generate half of those sounds.

The random object generates numbers from 0 to its right argument every time it receives a bang, so if you use this number to control the frequency of a sound generator, it can definitely play a different note every time.

It would kind of help to see your patch for this, because I'm not sure where you are going wrong.

Hmmm it seems that using asio and my Tascam US-144 interface I get a minimum of 25 ms latency (I've been using windows 7). I wonder if my interface just isn't fast enough for that or if it's an issue that could be fixed with more tweaking.

I haven't been using JACK, but I guess that wouldn't really make a difference because it just utilizes other drivers like asio.

I'll consider using Fedora or maybe some other variant of Linux; I think it might be nice having audio stuff separate from my windows boot up anyway. I haven't ever tried installing another OS and dual-booting, but I guess it can't be too difficult.

As a guitarist and Pd user I really appreciate this blog! One thing that caught my eye was the section on using Linux as an operating system. You mentioned that with Windows, you found it nearly impossible to run with a reasonably short latency.

I've noticed the same problem; if I try going below around 50 ms, I only get terrible noise coming out of PD. Did switching to Linux really get you to around 5-6 ms of latency? I didn't know this was even possible in PD, and if it means being able to process guitar in real time, maybe I should consider installing Linux.

Really great sounding stuff! Thanks for sharing! I especially like the 12TET video; I find the that the overall tonal structure in that one is something I can actually follow musically, where the others are a little bit more difficult to follow (but still interesting).

Is all of the synthesis done in PD, too? The quality of your sounds always surprises me; I would expect these kinds of sounds from hardware or pricy soft-synths, but actually developing instruments that sound that good takes quite a bit of dedication and a deep understanding of sound. The mix also sounds very nice.

I feel like even though these kinds of patches are really fascinating, they don't get shared too often on these forums, so thanks again for posting!

Wait, so all of the granular/crazy spectral ghostly sounding stuff is from Soundhack externals? I really need to explore those more.

And is the guitar recorded direct into a mixer or interface, or are you sending it through amps or something first? I guess if you are recording separate channels for each string, you probably are just going direct, but I find it insane that the guitar tones here are developed through simple processing.

Anyway, I really feel like I can't compliment you enough on this latest work. The sound is phenomenal, the harmonic ideas are pleasing and interesting, and the actual guitar playing retains its musicality even post-processing. The fact that you can do all of this and even make it work in a live setting is mind-blowing to me.

Wow, I really love this! It's simple enough that I can follow what is going on but the sound is beautiful and easy to get good results with.

I'm not sure how I would change the patch, but I could see using the sound as a basis for a track. If I ever make anything new out of it I'll be sure to post it here.

Haha it's good to see you answering your own questions.

Well if you have a slider that is sending values from -1 to +1, you should just be able to add a multiplication object

• after the value to get a range from -50 to +50.

I am in a similar position to you; I just finished up my first year of college and I'm trying to become more comfortable with PD and audio processing in general. I only have taken a semester of calculus, so the math needed for DSP (linear algebra, differential equations, etc.) is all stuff that I am trying to teach myself (it's not happening very quickly).

The thing that I will say, though, is that the most interesting results you will get in audio are probably from experimentation, though. Knowing the math will help you do some crazy, professional sounding things with patience, but as mod said you are going to learn a lot by just trying new things and exploring patches.

Some really great results can come out of fairly simple processes. For example, I am trying to process guitar arpeggios that I recorded in different ways. So far I have sampled the original recording at normal, half, and double speeds and layered them together, then convolved them with white noise (I used [partconv~] it sort of blurs the different frequencies together and can create a nice drone-type sound).

Finally I played this audio back in a new patch and filtered it with [vcf~] objects tuned to different notes in the original arpeggio and triggered on and off with a simple sequencer. With a little delay and reverb, I ended up building a sound I really like and can probably use as the basis for a track.

I guess my point is that I barely have any idea what I'm doing, but I am enjoying myself and learning new, potentially useful techniques. I think the fact that you are already working with PD and reading pretty advanced material shows that you have passion and are learning quickly.

Sorry if I'm not really helping, but just realize that if even you only understand half of the logic behind this stuff, someone else like me only understands a quarter. Keep reading, asking questions, and experimenting.

This might not be too useful for actually using PD and learning all of the applications of FFT, but if you do want to know more of the math behind FFT you can read through this book: https://ccrma.stanford.edu/~jos/mdft/mdft.html

It only assumes that you have a strong knowledge of algebra and maybe a little calculus. I'm reading through the book now, and although I do have a mathematical background up to first year calculus, I do find it extremely confusing at times. Still, it's worth checking out.

I really like this synth. Good job! I don't really understand the theory behind it, though; I don't know what the cosine waveshapers do or what is going on in the big filter section. Do you know anywhere I can read up on some of the theory? I am only just starting to familiarize myself with some basic DSP mathematics. Maybe I just need to spend some time learning basics before I try to comprehend this.

Dkeller, I tried this synth with a midi keyboard, but I also tried it out with your guitar patch. Nice job with the patch; it does a good job of detecting the notes and turning them on and off. The only problem with using a guitar with a patch like this is that the guitar is obviously monophonic while the synth is polyphonic. I guess it would be nice in this situation to have one of those hexaphonic pickups

I agree that the lower partials are a little too quiet. If I play a lower note on the keyboard I can barely hear it but when I start rolling the mod wheel a bit the higher partials come in and they are a lot louder. It sounds great already but it would be nice to hear a bit more of the droning bass sound.