R/E/P Community

Please login or register.

Login with username, password and session length
Advanced search  

Pages: [1] 2 3  All   Go Down

Author Topic: Summing  (Read 20955 times)

dobster

  • Full Member
  • ***
  • Offline Offline
  • Posts: 152
Summing
« on: October 17, 2006, 10:29:04 pm »

i've been seeing some arguments that sending tracks directly to a/d outputs from within a DAW sounds/sums better than sending tracks to a bus or master bus and then to the a/d output.
isn't it all 1's and 0's? any validity to this?
Logged

danlavry

  • Hero Member
  • *****
  • Offline Offline
  • Posts: 997
Re: Summing
« Reply #1 on: October 18, 2006, 05:51:56 pm »

dobster wrote on Wed, 18 October 2006 03:29

i've been seeing some arguments that sending tracks directly to a/d outputs from within a DAW sounds/sums better than sending tracks to a bus or master bus and then to the a/d output.
isn't it all 1's and 0's? any validity to this?


Did you mean to say DA (instead of AD?)

Regards
Dan Lavry

Logged

dobster

  • Full Member
  • ***
  • Offline Offline
  • Posts: 152
Re: Summing
« Reply #2 on: October 18, 2006, 07:31:45 pm »

yes dan, absolutely. my mistake..apologies. I meant D/A, thanks for the oversight

i've been seeing some arguments that sending tracks directly to d/a outputs from within a DAW sounds/sums better than sending tracks to a bus or master bus in the DAW and then to the d/a output.
isn't it all 1's and 0's? any validity to this?
Logged

danlavry

  • Hero Member
  • *****
  • Offline Offline
  • Posts: 997
Re: Summing
« Reply #3 on: October 19, 2006, 01:43:54 pm »

dobster wrote on Thu, 19 October 2006 00:31

yes dan, absolutely. my mistake..apologies. I meant D/A, thanks for the oversight

i've been seeing some arguments that sending tracks directly to d/a outputs from within a DAW sounds/sums better than sending tracks to a bus or master bus in the DAW and then to the d/a output.
isn't it all 1's and 0's? any validity to this?


The question itself is about subjective opinions and tastes - what "sounds better". I can not answer it in a subjective manner. I can see a lot of explanations as to why one may prefer the summing done in the analog world. They may like a particular sound of a DA, an analog mixer, and AD... I also heard some people preferring the smooth response and feel of the analog sliders (compared to on screen sliders and knobs).

If your question is strictly about summing, then it is difficult for me to see any validity to arguments against doing it in the digital domain. Once the data is in the digital domain, summing in digital eliminates the need for additional DA, analog mix and AD processing, which can only cause more deviation from the original waveform.

But if the question goes beyond summing, and some of the processing is done in analog (beyond just summing), then we are getting into comparing analog processing to digital processing. As a rule, linear processing (such as summing, EQ, re-verb...) is well suited for digital work. At the same time, any processing that calls for non linearity (such as compression, limiting, tube emulation...) may be better in analog, because a non linear processing in digital generates alias energy, which is very non musical. I am not condemning all non linear digital processing, but I am suggesting that it needs to be "carefully evaluated". While compression in digital may or may not be OK (implementation dependent), hard limiting in digital is probably bad news no matter what you do to try and fix it...

But even the above comments may not stand up to "what sounds good". For me, high degree of aliasing is always real bad news. The alias energy, unlike harmonics of musical instruments, does not fall on frequencies that are multiple of the fundamental pitch. So it sounds horrible to me, but in fact even that is a subjective opinion.

Regards
Dan Lavry
http://www.lavryengineering.com        
Logged

dobster

  • Full Member
  • ***
  • Offline Offline
  • Posts: 152
Re: Summing
« Reply #4 on: October 19, 2006, 04:16:28 pm »

Dan thanks for the response

I know my question is easily confused with "digital summing vs. analog summing: which is better?" But, I'm actually not concerned with summing "outside the box" but whether or not summing tracks through bus's ITB is a good idea or not. And by that I mean, should bussing be kept to a minimum? Some of the advocates for NOT using bus's in the DAW claim that the audio sounds "mushier" and/or smaller than simply letting the tracks  in the DAW sum to the D/A (the track outputs directly to an output). Is there any validity to this? Does that process even exist? In other words, is the summing happening in the DAW at all times anyway when one remains digital? There is no summing at the converters? Just, conversion to analog for monitoring?

You have though, in your response, sparked another question that I might as well ask now.

You, having said this...

" At the same time, any processing that calls for non linearity (such as compression, limiting, tube emulation...) may be better in analog, because a non linear processing in digital generates alias energy, which is very non musical."

I'd like to know then, how would you feel about the idea of recording the output of a digital track in the DAW with some type of non-linear processing plugin in realtime? I guess I'm asking, do non-linear processes only present alias artifacts when processed digitally and rendered? Or do they also present those artifacts while audio is playing through the plugin listening to it in realtime? Regardless, if so or if not, is there any benefit to letting audio play through the plugin and capturing that realtime playback back into the DAW? Assuming of course and putting aside the question are the converters worth it, etc. This is directed strictly towards the process in and of itself
thanks
Logged

ruffrecords

  • Full Member
  • ***
  • Offline Offline
  • Posts: 130
Re: Summing
« Reply #5 on: October 22, 2006, 12:41:55 pm »

dobster wrote on Wed, 18 October 2006 03:29

i've been seeing some arguments that sending tracks directly to a/d outputs from within a DAW sounds/sums better than sending tracks to a bus or master bus and then to the a/d output.
isn't it all 1's and 0's? any validity to this?

Some. Going direct means essentially the bits are sent directly to the D/A. Going via the bus means the bits get multiplied by the fader value. Assuming this value is 1 then it depends on the DAW as to what happens. A smart DAW might realise x1 means do nothing and it would do nothing. Other DAWs might just do the x1 multiplication. Depending on the DAW this might or might not give exactly the same output as the direct connection. It depends on how math is handled in the DAW and whether or not it has any rounding errors. This means some samples may not come out at exactly the same value as they went in.

Ian
Logged

dobster

  • Full Member
  • ***
  • Offline Offline
  • Posts: 152
Re: Summing
« Reply #6 on: October 22, 2006, 09:56:53 pm »

hey Ian...thanks

so, how does one quantify whether or not a DAW possesses there rounding errors or not? and if that bus fader is at "unity" then is there much of a difference then going directly to the D/A directly?
Logged

Jon Hodgson

  • Hero Member
  • *****
  • Offline Offline
  • Posts: 1854
Re: Summing
« Reply #7 on: October 23, 2006, 05:04:24 am »

dobster wrote on Mon, 23 October 2006 02:56

hey Ian...thanks

so, how does one quantify whether or not a DAW possesses there rounding errors or not? and if that bus fader is at "unity" then is there much of a difference then going directly to the D/A directly?


Any DAW will have rounding errors

HOWEVER, in setting the level and summing (as opposed to say, EQ) the process is so simple (one multiply and one add per channel) that the error is only going to be half a bit maximum. You are not going to hear a half bit error in a 24 bit system.

Logged

danlavry

  • Hero Member
  • *****
  • Offline Offline
  • Posts: 997
Re: Summing
« Reply #8 on: October 23, 2006, 04:00:15 pm »

dobster wrote on Thu, 19 October 2006 21:16

Dan thanks for the response

I'd like to know then, how would you feel about the idea of recording the output of a digital track in the DAW with some type of non-linear processing plugin in realtime?
thanks


Aliasing occurs the instance one tries to include energy (signal) containing frequencies above half the sample rate (at a specific point in the circuit).
To avoid aliasing when converting with an AD, you must make sure that there is no energy there at frequencies above Nyquist (half the sample rate of the converter input circuitry - called the modulator). You can do it by pre filtering (analog filter) the high frequencies, by operating the modulator at higher rate (oversampling) to a high enough rate.... In practice you get to do both (filter and fast operating modulator).

You can do all sorts of processing in analog, and there will be no aliasing. Non linear process will generate high frequency, which when converting to analog can be dealt with as stated above by:
1. Pre filtering
2. Faster rate of conversion (oversampled modulator).

But when you are in the digital domain, as soon as you did a non linear operation, the aliasing is already there, and whatever falls on the audible range can not be removed, it is too late, because the non linear process and the aliasing happen simultaneously. You can help it some by doing the process at high oversampling rate, but whatever does alias to the audio, stays there. You can not filter it before the fact, you can not do it after the fact.

So you can do non linear processing in analog. If you want it in digital, make sure that the level of aliasing is acceptable.

Regards
Dan Lavry
http://www.lavryengineering.com
Logged

dobster

  • Full Member
  • ***
  • Offline Offline
  • Posts: 152
Re: Summing
« Reply #9 on: October 23, 2006, 08:53:10 pm »

thanks Jon and Dan...

jon, just to clarify, what if you bus @ unity then? meaning, no level adjustment at the bus?

dan, is it possible to create a plugin that upsamples momentarily locally on a track to process and then downsample again? or is that just a dumb idea?
Logged

danlavry

  • Hero Member
  • *****
  • Offline Offline
  • Posts: 997
Re: Summing
« Reply #10 on: October 24, 2006, 02:30:55 am »

dobster wrote on Tue, 24 October 2006 01:53

thanks Jon and Dan...

jon, just to clarify, what if you bus @ unity then? meaning, no level adjustment at the bus?

dan, is it possible to create a plugin that upsamples momentarily locally on a track to process and then downsample again? or is that just a dumb idea?


Not dumb at all. That is the best way, and maybe the only way to overcome the aliasing. The problem is that depending on the nature of non linearity, you may have to up sample to 10MHz or beyond, and that is heavy computational requirement.
Why so fast? It is true that as a rule, the overtones made by the non linearity decay as you go tom higher frequency, but when we impose a logarithmic curve on the harmonic decay, it does not decay fast enough. The ear is nearly logarithmic in terms of response to amplitude, so a decay to say 1% is really only "40dB down", and even a decay to .1% is only 60dB down....

So one needs to evaluate how much aliasing is there on a "case by case" basis. As a rule, I think that a real "hard" non linearity, such as a digital hard limiter, would be nearly hopeless. I suspect that a non linearity of say a digital compressor may be handled well, because most of the non linearity is at very slow speed (the envelope)....

Regards
Dan Lavry
http://www.lavryengineering.com    
Logged

ruffrecords

  • Full Member
  • ***
  • Offline Offline
  • Posts: 130
Re: Summing
« Reply #11 on: October 24, 2006, 05:57:27 am »

Jon Hodgson wrote on Mon, 23 October 2006 10:04


Any DAW will have rounding errors

HOWEVER, in setting the level and summing (as opposed to say, EQ) the process is so simple (one multiply and one add per channel) that the error is only going to be half a bit maximum. You are not going to hear a half bit error in a 24 bit system.

The internal word length (number of bits used) will be longer than the required final answer. As you say, a maximum half bit rounding error may occur when this is converted back to the required word length.

There is another source of error though and that is in the math itself. Some DAWs use integer math but with large word lengths (as much as 56 bits) and others use a floating point representation of the data. The results form these will themselves contain small errors but usually they are second order compared to the rounding error in converting back to the required word length. However, even some simple multiplications where the fractiional part of the answer is close to o.5 bit will occasionally have a small error which casues the subsequent rounding to go in the 'wrong' direction giving occasional one bit errors.

Ian
Logged

dobster

  • Full Member
  • ***
  • Offline Offline
  • Posts: 152
Re: Summing
« Reply #12 on: October 24, 2006, 12:44:14 pm »

Dan, excellent clarification... So, what i gather is, a good solution to digital processing of non-linearity overall is to either use outboard analog gear to perform those operations assuming great converters though. Or, and forgive me if this still isnt the right idea, use a quality upsampler, export the tracks to the upsampler using great converters, once again, and then into the DAW to process? But this as you said, still requires alot of CPU right? Because then the local sample rate must change to match those newly converted tracks.

damn it, somebody make that upsampling plugin!  Rolling Eyes


So Ian, thanks for that and as I understand it then, floating point should be more superior to fixed, no? And even better is to use a high native bit depth and on top a high floating point for processing within that DAW
Logged

danlavry

  • Hero Member
  • *****
  • Offline Offline
  • Posts: 997
Re: Summing
« Reply #13 on: October 24, 2006, 02:00:57 pm »

dobster wrote on Tue, 24 October 2006 17:44

Dan, excellent clarification... So, what i gather is, a good solution to digital processing of non-linearity overall is to either use outboard analog gear to perform those operations assuming great converters though. Or, and forgive me if this still isnt the right idea, use a quality upsampler, export the tracks to the upsampler using great converters, once again, and then into the DAW to process? But this as you said, still requires alot of CPU right? Because then the local sample rate must change to match those newly converted tracks.

damn it, somebody make that upsampling plugin!  Rolling Eyes


So Ian, thanks for that and as I understand it then, floating point should be more superior to fixed, no? And even better is to use a high native bit depth and on top a high floating point for processing within that DAW



Hi again,

First, regarding your comment about a loss of 1/2 bit at the final truncation:
It is best to have that final truncation done with dither. At first glance, that 1/2 least significant bit may not "look as bad". But it is worse then it looks, and here is why:

If you add dither, the overall noise energy will be a little higher, but it will be "spread out" evenly across the spectrum.

If you do not add dither, the overall noise will be lower, but you may end up with "spikes on the FFT" - concentration of energy at harmonics frequencies of the tones. A tiny sine wave, when quantized by say 2 levels only, is in fact a square wave, thus it has a bunch of odd harmonics... It also sounds like a square wave... And also, without dither, there is "noise modulation" - the noise floor changes with the music (instead of being constant). As a rule you will not hear the lack of dither at a 24 bit level, so the above may not be so impotent, but it is a good form to dither the truncations.

Regarding the "upsampling plugging". Again, as a rule we are not talking about up-sampling to 192KHz or 384KHz... An up sampler for say "hard limiter" (very non linear operation) may call for up sampling at 10MHz or more. I heard of such stuff being done for some speaker project. The problem is far greater AFTER the up sampling. Doing DSP on say 100Hz sine wave sampled by say 44.1Khz is in fact dealing with a ratio of 441 to 1 (sample rate to signal). In other words, it takes 441 data points to "express" a single cycle of 100Hz. Now lets try to store 1 cycle of 100Hz with 10MHz sample rate - it would take 100000 samples to do that!

What I am getting at is a general comment: the required DSP power grows very fast as you go to higher sample rates. The growth is often far beyond linear. The very brilliant Dr. Richard Cabot did a great presentation on that subject in an AES workshop I was chairing about 2 years ago.

Regards
Dan Lavry
http://www.lavryengineering.com
Logged

ruffrecords

  • Full Member
  • ***
  • Offline Offline
  • Posts: 130
Re: Summing
« Reply #14 on: October 24, 2006, 05:19:22 pm »

danlavry wrote on Tue, 24 October 2006 19:00


It is best to have that final truncation done with dither. At first glance, that 1/2 least significant bit may not "look as bad". But it is worse then it looks, and here is why:

If you add dither, the overall noise energy will be a little higher, but it will be "spread out" evenly across the spectrum.

I agree any final truncation should be done with dither for the best sound. The interesting word though is "final". I suspect 'final' applies to any mix within the DAW for example just after a bounce. If it does then it implies noise will build up with successive digital bounces just the way it used to in analog days.

Ian
Logged
Pages: [1] 2 3  All   Go Up