Analog summing emulation idea

[farlukar]

Clearly some people dont understand what I am talking about or are clearly ignorant and inexperienced.

if you're talking about

I dumped hardware in 2004 to go completely native but found that my creativity went down the toilet!!!

Yeah... always blame the tools

If it should be obvious that a slightly different sound colouring kills creativity - colour me ignorant

[Kingston]

ahh, finally this thread turned funny... after some needless wanking by yours truly.

[kritikon]

The two experiences of making music on a DAW+host and making music through a desk are worlds apart. Possibly there are some subtle audible differences, and undoubtedly there are varioius ways of recreating the differences in the digital world. I used to have immense fun with all my h/w and big mixers. At present I'm only on a PC with one measly keyboard...and I f**king hate it. Soundwise...not an awful lot of difference really...the advantages of s/w outweigh the disadvantages (IMO). But...soon I'll have a new studio organised and built, and all my old h/w will come out, and I know without any argument that I will have immeasurably more fun making music. Every single outdated piece is coming out and will get used, even if the sound is worse than even freebie s/w.


And that's what the audio-wankery people need to separate out...the experience of making the music and the effect that has on the experience listening to it. If you hate making it, you're most likely going to hate listening to it. Conversely I occasionally listen to some of the stuff I did years ago - made on 303s, OSCars, Jupiters, 909s, all analogue desks, rack units with big swirly knobs and banks of flashing lights - I had a f**king ball making it, and consequently I tend to enjoy listening to them. But when I really listen objectively, I notice the overpowering wobbly bottom ends, the complete lack of any clarity compared to even my first DAW mixes, the reverb mushes (even on subtly-used reverbs) and almost no sparkle without using a BBE on number 11...some of them are so syruppy and swirly, as I had to overdo the BBE to negate high-less mixes that they are almost laughable)...(and by the way...you're both wrong about BBEs - check their literature...I have the manual right here in front of me )


Of course your musical creativity dies with a PC and a mouse. It's not because of the sound, it's because it's a f**king tedious disconnected way to make music. Digital doesn't kill your sound...not interacting with your instruments kills your sound. You don't have to be a concert pianist to interact with an instrument - I play really badly, but I know for a fact I get better riffs when I patch up a real live analogue synth with lots of knobs to twiddle, because I cater the riff to the sound I'm patching up - I'm involved with it, rather than sat in front of a screen disconnecteing the riff from the synth. I can get tedious predictable riffs equally with s/w and h/w - put me in front of a PC screen and I can give you any number of tedious tunes that make you sleep I can do that equally well in front of a Wavestation too - another GUI that gives no interaction whatsoever - being h/w and s/w makes no odds to me...it's the interface. Being an analogue synth doesn't necessarily make it better than a digital h/w synth or VSTi - what makes it infinitely better is there's a knob for each and every parameter. What makes the difference on a big desk is that you actually play the mixer as the song progresses - you interact with it.


Someone like the Mad Professor could probably technically make an equally good mix with all s/w and a PC screen. What makes some of his classical fluid older mixes is not that his sound is any different on an analogue desk, but that he rides the faders and FX knobs as he goes on a desk - he plays the mix rather than computes it. I used to do that...I used to constantly tweak a mix as I went - subtle changes here and there, a little roll off on the bass in the main bit, a little more reverb in the breakdown, a tad of delay on the kick every now and then to stop the boredom. I don't do that know - it would reduce the boredom of my mixes, but it's too f**king boring to actually do it on my method of mixing (s/w & screen). It's a catch22. I can make crap mixes, not because I'm stuck with digital rather than analogue (believe me I used to have a serious amount of h/w once-upon-a-time), but because I'm stuck with a crap interface.

My best investment would be a big mixer again, with a huge soundcard with multiple outs - I've delayed and delayed though, because there're other goodies that are so cheap in s/w that I want to buy first. But that one purchase would improve my music. It could actually be a controller in all digital (I haven't seen one even remotely good enough to equal a proper mixer yet, at the price I can afford...) even if it was 0s and 1s, it would improve my sound, simply because I'd get more involved with my mixing again.


Analogue/digital/summing/winkywankywoo.
It's all irrelevant compared to other stuff.

[aciddose]

its just all about variation. the problem is, the analog equipment does this by itself. you cannot turn it on or off. in digital equipment, you need to do variation intentionally, you can turn it on and off, but it is going to take quite a large amount of processing power to produce a simple eq-distortion-noise-panning-amplitude adjustment for every channel of audio individually.

in my opinion, it is actually easier to get the 'analog mixer' effect by actually using an analog console. go down to a music shop and buy one for a couple hundred dollars. make sure to get an older console (late 70s, early 80s) and make sure not to get one of those 'mini' consoles made of plastic. you should be looking for the wood/steel construction and the discrete circuits made with through-hole components.

using a good card with multiple ins/outs, you can simply run all your channels out through the mixer rather than having them mixed digitally, and take the output of the mixer onto the final bus input in your host. i've tried it, it works beautifully. it is a good amount of effort, but considering the quality of the effect compared to what the quality might be if done digitally, i think it is worth it.

of course, you would have to remember all analog mixers are going to sound quite different. i own a "audio pro 16" made by traynor around 82. this is a pretty good console for fooling with. the newer and more expensive consoles are usually in my experiance going to sound closer to the digital 'clean' sound, so if using the console to get the 'analog' sound it would be best to use the cheapest oldest peice of junk you can find.

apart from that, it is also fairly easy to create your own analog mixer or inserts that can adjust the sound of your signals.



this is a fairly basic circuit that can be used for experimentation. the output might be best if buffered, i would probably use a jfet follower with 12k from the rails.

it is easy to use this to create a multichannel / multibus mixer with features like eq.

the amount of real coloration from such a mixer is minimal in most cases. in building it myself though, i was surprised at how i really can tell the difference between an unmodified signal and one routed through this amplifier. this is an asymetric amplifier. if you believe an opamp mixer makes much difference to the sound.. well.. good luck. i wouldnt bother with non-discrete circuits.

[friteuse]

kritikon, so true! Another thing is that since DAW, we all became undo junkies, what prevents us from taking the trouble to make things right just from the beginning. Same with copy&paste...

However, I am one of those, who enjoyed the work with mouse, keyboard and crt from the first day. It awakes in me kind of same enthusiasm, like h/w for you.

But that's not the point. It's only about analog summing. For me, it's not better than digital summing, and I'm looking forward to the time, when apple makes Logic 64 bit...

[Aleksey Vaneev]

friteuse, from my experience, D/A and A/D add a bit of harmonics of psychoacoustic levels. You can't say they are present for sure, but they do change the overall perception. Small noise levels do that as well - noise masks information and then your brain or heart may build up a more positive impression due to absence of precise sonic information. Anyway, due to such differences comparing DAW output to analog mixing box's output may not be objective.

I believe todays paradigm is to produce such sound that is spectacular at high digital definition - this includes synth instruments, micing paths. 'Sterile'-precise DAW is a requirement. If digital recording sounds lousy, it means sound sources are lousy in the first place. I know some people may argue about this, but good digital does reveal true sonic quality. You may actually hear that drum set you've recently miked sounds like sh*t and it sounds so not because you've used 'vintage' equipment, but because you've used wrong mics, wrong preamps, wrong technique and... wrong drums. In digital, every component is important.

[camsr]

64 bit is good. 192,000hz is GREAT.

At the higher sample rate, digital recording can catch what a fully analog system outputs. Think about this... when your system is fully analog front to back, your max frequency is limitless. Now you record this signal, and the sampling turns frequencies in the 11000 - 24000hz range into mush. With the higher sampling rate, waves in the 24000hz region (which I consider MY max hearing range) are represented with 8! samples each! NOW recording your analog input is going to be far more accurate because you can catch the micro-phases of the signals. With more samples, frequencies can be better represented in the time domain (phase). And it sounds different.

And Im seriously done trying to be smart

[Aleksey Vaneev]

camsr, sample rate is not as important as Hardware and Software dealing with it. 44.1kHz represents frequencies up to 22050 Hz precisely, including phase information. You will be surprised that from mathematical point of view there's no sense in having 8 samples to represent 20 kHz signal. It is simply redundant.

You must be joking you hear frequencies up to 24kHz... Stable threshold is 16 kHz. I personally can say that I hear up to 20kHz - but that's at a very high speaker volume (sinewave test). And I can't be sure this hearing is quantitative - due to spectral masking in most cases you won't hear so high frequencies in any actual recording.

[Kingston]

With the higher sampling rate, waves in the 24000hz region (which I consider MY max hearing range) are represented with 8! samples each! NOW recording your analog input is going to be far more accurate because you can catch the micro-phases of the signals. With more samples, frequencies can be better represented in the time domain (phase). And it sounds different.

*sigh*

one word camsr: 'nyquist' The only reason it sounds different is due to the quality of the AD/DA and clocking. The high end luxury models have practically no difference between 96khz and 44.1khz.

and a little remainder, you are not a dog.

[farlukar]

Being able to hear up to 24 kHz - not very likely but also not impossible, certainly at younger age.
But I can hardly imagine it making a lot of difference in a piece of music OTOH.

[Rangtangtang]

Can you hear insects, butterfly wings, spiders and sound from outerspace. Dogs can hear high frequencies.

[scam_artist]

(comments here)

great points krikiton, I think you've nailed the core issue with producing music on a computer - the hands on feel isn't there yet, despite how many midi controllers we have, because the midi controllers can't comfortably represent every software synth we have in terms of layout (midi controllers have a fixed physical layout, for those who missed the point - despite the fact that you can reassign controllers, all the knobs/sliders are still physically in the same place!)

[irtimid]

Aleksey certainly knows his stuff. I hadn't read his ideas on this before.

But it seems very obvious to me now. Analog components definately come in tolerances - and even 1% is fairly random.

Imagine your stereo signal runs through a few 100K ohm resistors - each of which could be anything from 99K to 101K ohms. The variation is voltage going to be significant.

Imagine a typical DAW mix with many tracks panned dead centre. Now mess up the pan of each one slightly. Then apply a dual band eq to each stereo track, and apply slightly different eq changes to each side.

The mix is going to sound like it has more space and depth. This is exactly what will happen if expensive analog summing is used - because component tolerances will effectively do just that.

Sure, the extra noise and distortion helps. But maybe that's not the major part of the sound? I think this random assymetrical tolerance thing might be a good thing.

It wouldn't add much, if any, overhead to a DAW to implement this analog-like behaviour. A user control knob, giving anything from clean digital to dirty analog would be cool. I could imagine a panel giving user control over the following:

noise
distortion
crosstalk
asymmetry - pan
asymmetry - eq
asymmmetry - phase

Could be interesing .. imagine a DAW mixer with presets for various digital or analog desks ... Neve, SSL, Mackie, Behringer, Paris, etc ..

Is there such thing as a dual band EQ, who can equalize Left & Right side seperate? if yes, which ones?

[C00kie]

Sorry I have to go back to the start of this:

it turned out that my Mackie mixer (or possibly the Audiophile card) colors the sound slightly differently left and right. If I swap the cables around, I hear the difference in the other ear. (Phew).

My guess is this can be explained by either slight volume difference (check trim & fader settings on the mixer, align left & right levels not on the VU meter of the mixer, but the digital meters in your DAW) or the EQ section (all pots clicked in the center?)

I say this because I did some actual measurements with RMAA on my Mackie unit. Although it sure is not "perfect" like digital, the difference between channels is neglectable in terms of audability. Well, on my unit anyway.

Sure the analog harwdare may colour the sound, but don't overlook "human user error"

[friteuse]

Being able to hear up to 24 kHz - not very likely but also not impossible, certainly at younger age.
But I can hardly imagine it making a lot of difference in a piece of music OTOH.

We do often forget, that we don't only hear frequencies as notes, but as phase (or location information) too. Recent experminets showed, that we are able to perceive run time differences (uh, is this the right word in engl.?) of 5-10 µs between left and right ear, corresponding to a freqeuncy of 50-100 kHz (see german "Studio Magazin" 12/05). So, we cannot hear a note at this frequency, but still an information, namly the location of a sound source. This information gets lost when working with sample rates < 200 kHz, so 192 kHz should be a good choice to leave most of them in the source signal...