Has anyone else noticed this? Isn't it a bit strange? I wonder if it's hardware specific and due to that some speakers work differently in the two directions so to say. Or do healthy ears actually notice a difference between the two?
I sometimes imagine i can hear differences from inverting the phase of the master bus in a DAW but iv'e always just shrugged it off as some artifact or auditory illusion or whatever, but this phenomenon just keeps bugging me. Or are my ears shot and i pick up pressure better in one direction? Or is this some artifact introduced by pure data that has got me confused?

Help me find my mind

@Booberg Well a sawtooth "blows" the speaker forwards and then it rapidly returns before the next "push" (if your speakers are in phase) while an inverted or reversed sawtooth "sucks"......

As @old says below, air resistance, hysteresis, mechanical overshoot, back EMF and much more (even for electrostatics) will compromise the final translation of a wave to air pressure.
What you see is not what you get.
David

@Booberg Yes, I definitely hear it, and I've posted on a similar thing which I thought you weren't supposed to be able to hear (the relative phase of saw partials). And it's not just subjective--I built a double-blind randomized test which I consistently pass with a high confidence level.

@Booberg The real world is not ideal and sawtooths are tricky things, your suspicions are on the right track. Analog electronics and our electro-mechanical speakers are full of faults when it comes to very quick transients like we have on one side of a sawtooth waveform, an instantaneous jump from 0 to 1 will slew as if you have a small amount of glide on the waveform, when it reaches 1 it can overshoot and go briefly to 1.1 or ring and cause a few cycles of low amplitude oscillation at a higher frequency. The response of this is often different when going from 0 to 1 vs from 1 to 0 and be exaggerated by DC offset inherent in the analog system which could cause only one side of the waveform to be slightly clipped. On top of that these effects increase with age and this is much of the "warm" sound of vintage equipment, especially speakers, their tired old cones and surrounds increase slew which results in a rounding/softening of waveforms and is also the cause of the expense of studio grade equipment, they are doing everything they can to minimize these effects which means lots of high precision components and complicated circuit topology. And then we can also add in room effects and our ears, both which will contribute but it is not so much a "healthy ear" as you suggest, no matter how healthy your ear is the upper limit of your hearing range decreases as you age, the young can hear higher pitches then the old. This is not a huge deal since the high frequency content affects the low frequency content and we do hear those effects and our brains fill in missing information that experience has taught them should be there, your upper limit needs to drop a fair amount to actually hinder your practical ability here, you just hear it differently than a young person but you still hear it.

@jameslo I attempted to answer your phase question when you asked it but the answer kept getting out of hand with no end in sight, i was having problems being concise. If memory serves there was something off about your test which made providing an answer surprisingly difficult and after typing up a very long response I realized it could very well be wrong because of this and I lost motivation. I will review it come weekend.

Highly appreciate your answers I guess it's hard to find one definitive answer but I feel better about my assumptions at least.
An additional detail in my case is that I almost exclusively use in-ear monitors, i think that could make a difference as they probably face all kinds of resistances in trying to pressurize the ear in one direction and the body of the earbud in the other. Much like closed back speaker cabinets vs open backs.. My Genelecs are banished from apartment use but I think I have to do a comparison next time I have some time over in the studio
I kind of wish I had some old synthesizer that could produce both directions of sawtooth waves to see how they compare in an oscilloscope.

@oid slew is such a powerful phenomenon that is quite new to me. I've found that incorporating slew distortion can really bring a digital synthesizer that extra bit towards that illusive analog warmth It's just hard finding data on it.. I've been working on a "ladder filter" that has some burn to it from just clipping but maybe the slewlimit of the transistors could be what it is lacking. Time to look up some datasheets i guess
And speaking of which, I recently modified Else's "slew~" to limit with a atan function instead of hardlimiting. It doesnt make a huge difference but I imagine it "softens" the effect a bit.. I'm still a bit new to coding for pure data so i'm sure there is some dead code in there and whatnot left from the original, i wasn't planning on sharing it really. But in case that sounds interesting I'll share the source: softslew~.c ((Inlet 2 or argument is "gain" and just multiplies the input + divides the output. Things start to happen at around 25 gain with unity level in..))

Goodnight

@Booberg In ear monitors are not going to give you and accurate representation and will certainly exaggerate such effects, they are not designed for quality, they are designed to make sure you can hear what is going on on stage without the issues caused by stage monitors, don't have to worry about walking out of their beam.

Slew is one of the many things which is largely ignored by programmers so there is scant information from that standpoint, but any decent electronics book will have plenty on the subject and every opamp data sheet will have it speced. It is pretty simple for the most part and you can do a good job mimicking it with [lop~], just set the frequency to the amount of slew you want. This will not get you anywhere near the transistor ladder sort of distortion, especially the early mini/modular transistor ladder sort of distortion which is caused by the ladder loading down the output stage, this creates a highly frequency dependent distortion which varies with cutoff and Q. Later transistor ladders and the ARP variation have a much simpler distortion which is largely your standard clipping distortion and not as musical/natural but more versatile/less restricting. A few years back I designed a distortion pedal around the mini transistor ladder, this weekend I will see if I still have my notes and send them your way if I do, I did a fairly extensive study of it's distortion characteristics and the various sources, should help you. I probably have my LTSpice models for it all, so if you run windows or have wine installed I can send you those as well which will give you very detailed information regarding it all.

@oid Just to clear things up, I don't use in ear monitors intended for live use, I have a couple of different models from AKG, Bang Olufsen, and some big asian ones. But your point still holds i guess.
That work on transistor ladders sounds very interesting! I do my work on mac, are LTspice models not cross compatible?
A big difference between slew and recursive filters is that slew does not alter phase afaik, so one can add slewlimiting to the feedback of a filter without things getting to whack. My initial tests didn't show any revolutionary changes in adding slew between the poles, but it does give a nice effect in the feedback loop. If you have a favorite ladder implementation in pure data I would love to try it

@Booberg

slew does not alter phase afaik

Yes it does. Slew is also "recursive" by nature, i.e the output can only change so fast. The response is just not exponential. PD has a slew filter you can play around with [slop~]

@Booberg Slew rate in electronics does not itself alter phase (arguable) as it is a byproduct, but there are countless things in most circuits which do alter phase. We use capacitors to block the DC voltages between different stages of the circuit so the DC voltages which set operating points for those stages don't interact with each other and alter the operating point of the other, these capacitors form filters with the resistances around them, phase is a much more fluid thing in the real world and less perfect than in the digital world. Sometimes we design out these capacitors to avoid these effects and directly couple the various stages in a circuit but this creates a teeter totter effect, altering one stage affects the others so you have to adjust them all and rebalance everything. But slew rate is not an issue in something like the transistor ladder since its effects will be insignificant compared to the effects of the filter unless the the cutoff is all the way up and Q all the way down and even then would likely be lost behind the circuits distortion and noise.

I do not know of any really good software transistor ladders, most if not all mimic the circuit as a whole instead of the individual stages which make up the circuit, this largely removes the highly interactive nature of the the stages and the controls but it is more CPU efficient and makes the filter respond as most people expect a filter to respond. The filter has four major components from the audio standpoint, input amp, the transistor ladder itself, the recovery/output amp, and the resonance path, each of these tailors the frequency response and distortion characteristic of the overall circuit and each of these influences how the others do this. Generally the primary source of distortion is the output amp loading down the ladder, but we can also overdrive the input for a different sound and high resonance settings will overdrive the ladder itself, changing the filters frequency can completely remove the distortion effects of the output amp and make the other sources primary and they each have a very different sort of distortion. We have four distinct sources of distortion and tone shaping besides the ladder at the core and all but the input circuitry are affected to some degree by the cutoff control voltage since changing the cutoff changes how the ladder loads the output amp which changes how the resonance path drives the core and loads the output stage. I know of no emulations which bothers to model all of this, they seem to all fixate on getting that handful of stereotypical sounds and call it good because they know that the first thing most people are going to do is set it for one of those sounds and if it gets that sound they will be convinced of it's authenticity, most have not spent enough time with the real thing or done extensive analysis of the circuit, they only know those stereotypical sounds, not how it responds to the musician.

Edit: I don't know if any of this actually matters or if anyone would notice the difference between the simplified emulations and an accurate one, musicians tend to be more practical than engineers and the practicality of my musician side lost out to my curiosity in such matters long ago. The real thing does respond much more like an acoustic instrument than an electronic instrument, which is why the mini was the first big success in synths, but I am not sure that plays any role these days.

LTSpice may be available for OSX, I don't actually know now that I think about it, if you have wine installed the windows version runs perfectly there. Check their site, it is free. I do not believe you would be able to run its circuits with other simulators but I could be wrong, the file syntax might just be plain old Spice.

@oid I had a bunch to deal last week with so i'm just getting back to this. Prodding around with slew in the feedback of my ladder code ended up being quite weird and perhaps a bit to sensitive to dynamics. I don't fully understand why but it sets off a very strong and weird sounding self-oscillation at higher resonance if not carefully tuned. Sound good at some volumes but absolute rubbish at others.. I don't think it is that missing link i've been looking for.
Regarding the formulation of ladder filters i find that stacking 4 one pole filters gets closer to the real deal than for example using 2 biquads. Especially of we ad some "random tolerances" by offsetting the cutoff frequency of a couple of poles as well as introduce some slightly asymmetrical clipping between stages. But it is hard balancing these things I think the most important aspect is probably finding a balance between the clipping points of the feedback and the "filter". I'm actually surprised by how close one can come to authenticity with just a little clipping in the right places. It's just that illusive last 5-10% or so that is just bugging me endlessly

LTSpice is definitively available for OSX, it just looks a bit different. My guess is that chematics are allso cross-platform, but i'm not well versed in the software. Best of luck with your endevours!