Phase adjustment plugin????

[ebeam]

OK, so there is this little box put out by Little Labs (http://www.littlelabs.com/ibp.html) that allows you to continuously adjust the phase of a signal. I find this to be a great idea and a tool that could prove very useful. I was wondering if a DAW plugin exists that can do the same thing. It seems to me that the principle of the device is fairly straight forward and someone should be able to create a dsp version. Of course, latency and such would likely be very critical. Any ideas?

[zekthedeadcow]

..it's a lot cheaper to zoom in and slide the wav over to get it in sync

[ebeam]

Ah, yes I do that too, but its not the same thing. Time and phase ARE related, but delaying one track is definitely not the same thing that this box does. For example, if you use 2 mics on a guitar cab - one up close on the speaker and one a couple feet away - part of the reason it sounds good is the time delay between the two mics. If you shift them in a DAW, you negate that effect, but if you just change the phase of one of the mics you can compensate for comb fitering and such and preserve the time delay between the mics. There are also other problems with sliding tracks in a DAW (think about time and frequency relationships).

[ebeam]

Here is a visual example of what I mean by time/frequency relationships.

So, most musical sounds are made up of a combination of frequencies. When you slide track around in DAW, you are pretty much looking at the low frequencies, since they will dominate the visual representation of the waveform. In this example, you have one mic at A and another at B (with time/distance increasing from left to right). So, you shift one of the waveforms over in your DAW so the low frequency element lines up and is visually in phase but the phase relationship between Signal A and Signal B for the high frequency is very different and will not necessarily be in phase. What you need to do is change the phase across the frequency spectrum (it WILL be different for different frequencies)and the IBP box is supposed to do this.

BTW, I am by no means an expert, so if you find flaw in my logic feel free to point it out.

[ebeam]

Yes, time and phase ARE related but are definitely NOT interchangeable. The IPB box is not a delay. Read the website - he goes into more detail.

[ebeam]

Read the pdf - http://www.littlelabs.com/IBPMAN.pdf - it specifically addresses this issue.

Not trying to turn this into a pissing match, but the physics of sound just doesn't agree with what you are talking about.

The fact of the matter is that your ears ARE going to tell you what sounds good. This is not some magic fixit box, but a tool that can be used along with time delay, mic position, etc to get the sound you want. Its not necessarily used to make the phase relationship between the two signals exactly the same, but to give you the ability to change that relationship without delay or moving one of your mics from that sweet spot.

[zekthedeadcow]

heh

read page 7, paragraph 2, ...begins with "From first hand experience..."

This is for the analogue people and is essentially a delay... in the above memntioned paragraph it tells about a store that has only sold one and then had it returned because the user didn't care for it

[ebeam]

Well, you are right that adjusting the time can eliminate some of the effects and have a similar effect to adjusting phase. But the key to this problem is in your last sentence - yes you can correct the phase for some frequencies with delay (maybe the ones that are most important to your ear) and make it sound better but the fact is that there will be other frequencies that remain out of phase. I'm not trying to argue how useful the device is in practice (cause I haven't tried it) but I assure you there is a difference between phase and time. Here is a quote from Jonathan Little (the guy who makes the IBP) from RAP about this:

"The IBP analog phase alignment tool(in between Phase) uses cascaded passive all pass filters. It is not a time delay box. Phase is frequency dependent . You can be out in time but be in phase. Polarity when you swap 0 and 180 is the right term however most mixing desks use the term phase and most engineers I know use it when describing the button... hence the use of phase invert on the IBP polarity button. Scotts experimenting with all pass filters was probably with a single stage all pass filter which has limited usability. Reading these posts has really put the fire under my ass to get the manual done. Phase seems to be pretty misunderstood."

I think phase is a little misunderstood here as well.

There is a discussion on Prosound Web about this right now:
http://recpit.prosoundweb.com/viewtopic.php?t=3983

Zekthedeadcow:
Did you read the first line of the second paragraph under ??Phase?? where it says the IBP is NOT a delay line????

[Bigus Dickus]

From a physics standpoint, distance and phase aren't quite as directly related as just moving a mic to change the phase. After all, you can do a 180 degree phase swap without any distance change at all, and moving a mic will affect the phase of different feqeuncies differently.

As for shifting phase between 0 and 180 degrees though, that is done in analog by "lagging" or "leading" the AC signal, through capacitors or inductors. While it technically isn't quite the same as time shifting the waveform, since it is in theory possible to retain the original attack and yet nudge the phase of the AC signal, in practice the end result is nearly identical to a simple time shift.

If you digitally shifted the phase of a waveform without moving the attack, then you would by definition alter sonically the sound of the attack. This happens in analog phase shifting as well.

[edit] After reading the information in the last post, I think I understand a little better what you are saying. My above comments really are indicative of a single frequency signal, and as such are probably incorrect in the more general context. I hadn't thought of the effects of a natural multi-frequency signal. Yes, analog phase shifting, either leading or lagging the waveform, is strongly frequency dependent. Doing this to a signal will shift the time alignment of some frequencies relative to itself, that is... relative to other frequencies contained in that same signal.

So I would say it does actually make sense to ask about a plugin that could accomplish this digitally. Such a plugin would time shift the waveform using an algorithm that calculates the time shift appropriate for each frequency component. This would actually more closely match the real world effect of moving the microphone.[/edit]

[ebeam]

Bigus-

Right. Except the plugin wouldn't time-shift the different frequencies by different amounts, but phase shift them by different amounts. A 80Hz signal travels at the same speed as a 80kHz signal, so there should be no frequency dependent time differences given controlled conditions (ie an anechoic chamber, no reflections).

The phase differences between two signals at different distances from one source differs across the frequency spectrum, but this difference is entirely predictable. The IBP operates on this premise by treating the low frequencies and high frequencies separately. He then apparently tuned the device so that it works well for the most common applications. I think this is only the tip of the iceberg as you could theoreticaly (actually probably easier with DSP) develop a device that shifted the phase continuously (or at least relatively continuously) across the entire spectrum.

[Bigus Dickus]

Originally posted by ebeam
Bigus-

Right. Except the plugin wouldn't time-shift the different frequencies by different amounts, but phase shift them by different amounts.

I agree with what you are saying, but having a knowledge of analog circuit behavoir I know that capacitors and inductors are effectively time shifting the waveform. Certain events in time (such as attack and release) don't get completely time shifted, but their sonic characteristics are changed as a result. A steady state phase shifted waveform is a time shifted waveform. They are interchangeable. Natural sounds aren't steady state, which means you either time shift attack and release etc., or you alter those time-dependent events sonically. There's no other options.

In the digital domain, it would be easier to simply time shift depending on frequency, and let the attacks fall where they may. A sufficiently complex algorithm could keep time alignment on events and alter them sonically as necessary, but that would probably be more trouble than it's worth.

[pipelineaudio]

I have been dreaming of this friggin box! We saw a demo a few months ago.

This sucker, though you could do the same thing on a daw later, aloows you RIGHT THEN AND THERE to do tons of experimenting and getting different tones very quickly. A daw doesnt let you hear it sweep the way this baby does, and during the brief time it was in my hands, I was recording bussed mics to single tracks, cuz I KNEW it was good enough I wouldnt have to line it up later. I still did my headphone phase alignment trick, but this gave me a whole nother range...the price is a shame, if its still way hi like it was

[ebeam]

Originally posted by Bigus Dickus
I agree with what you are saying, but having a knowledge of analog circuit behavoir I know that capacitors and inductors are effectively time shifting the waveform.

Thats interesting. So this type of device just shifts each frequency (or band of frequencies I assume) on the order of the phase difference (for example, 90deg out of phase would be a time shift of 1/4 wavelength)? So, this thing is really nothing more than a multiband delay? It just notches out a group of frequencies and delays each band by a time related to the average frequency or something??....just guessing here.

Who says the phase shifted waves can't be altered sonically? I doubt you could do any of this without some distortion anyhow? Even so, some have said that humans can even hear the difference between signals of opposite polarity.

Anyhow, analog circuits aside, is it theoretically possible to change phase without changing time? Any physics guys out there? I believe its possible if you think about the fact that any wav is just a complex summation of sinusoidal waves with different amplitudes and frequencies/wavelengths. Just as a perfect sine wave can be phase shifted without time delay, in theory, you could do the same thing by breaking that waveform down (ie fourier transform) to the individual frequency/amplitude components, phase shift each of the frequencies independently and combine them again? The accuracy of such a program would depend on the precision of the fourier transform (and the amount of data, ie bit rate). So, latency may be a concern although I'm not sure how processor intensive that operation would be. I think you might even be able to analyze 2 waveforms and have the software match the phase of the two automatically.

[Bigus Dickus]

This is getting kind of deep here.

Originally posted by ebeam
Thats interesting. So this type of device just shifts each frequency (or band of frequencies I assume) on the order of the phase difference (for example, 90deg out of phase would be a time shift of 1/4 wavelength)? So, this thing is really nothing more than a multiband delay?

I can't claim to know the inner workings of this analog effects box without studying the circuit (and I'm not an EE by trade anyway), but I do know the basics of AC power theory, which is all analog sound is... a summation of AC signals.

The equations defining AC power manipulation are differential equations, and both time and frequency dependent. In short, if you solve a circuit behavior to predict a certain output when a given AC signal is run through a known capacitive or inductive circuit, you will find that the answer is dependent on the frequency of the input signal. This means that when a complex signal is used as an input, the analog circuit will affet each frequency component of that signal differently. It's not a "band" type of effect, but rather every subtle frequency can be though of as being split from the whole, ran through the circuit (which gives an output dependent on the frequency), and then summed with all the other component frequencies.

Anyhow, analog circuits aside, is it theoretically possible to change phase without changing time? Any physics guys out there?

It depends on your perspective. If you're talking about a pure sine wave, then answer is yes. All you have to do is time shift the waveform digitally, and then chop off or extend the attack to realign the start of the wave to the original starting time. Likewise for release.

If you're talking about a multi-frequency sound, then yes and no. You can use fourier transforms to seperate the waveform into all it's frequency components, do the above time shifting (which is phase shifting for pure waveforms) on each of the sine wave components, chop or extend attack and release to time align, and then recombine frequency components. The only problem with that is the alteration of the attack and release to time align will change the sonic character. It's unavoidable. For example, an attack starting in a trough will sound different than the original attack starting at 0db.

So, it's a matter of perspective. Can digital phaseshifting be done? Yeah, I guess, with enough computer power. It would probably take a while to calculate the transform at a high resolution for a long waveform. Can you do the phaseshitfing without altering attack and release? No. Not a big deal though, since you are doing the same thing in analog, and in any case the phase (time) shifted waveform will sound different in general, due to the different cancellations and interferences that will be generated from sliding frequency components in time relative to one another.

Just as a perfect sine wave can be phase shifted without time delay

A matter of semantics, perhaps, but I would say that phase shifting a perfect sine wave is time delaying (or advancing, take your pick) the waveform. Mathematically, it is identical, so long as you are away from attack and release points.

[ebeam]

Originally posted by Bigus Dickus
This is getting kind of deep here.

Well, FWIW I'm enjoying this....

EE definitely isn't my forte so I appreciate the input.

I guess it may be a matter of semantics. But think about this: What is the difference between a perfect sine wave and a perfect cosine wave (with the same amplitude and frequency) starting and ending at the same time? Phase. No time difference, just 90 degrees out of phase. You can calculate the phase difference between these two and redraw one to match the other without changing the attack and release at all. If you can do that, you can do the same for a more complicated wave form, it'll just take you longer.

And a fourier transformation is a method for solving/manipulating a differential equation...

[ebeam]

You know, I just thought of a great analogy for phase shift without time delay.

Think about the profile of a corkscrew spinning on a hard, flat surface. The corkscrew's heght doesn't change (time), the thickness (amplitude) and spacing (frequency) don't change throughout the length, just the phase.

[Bigus Dickus]

Originally posted by ebeam
What is the difference between a perfect sine wave and a perfect cosine wave (with the same amplitude and frequency) starting and ending at the same time? Phase. No time difference, just 90 degrees out of phase.

The definitions are completely interchangeable for sine waves. Time, phase, same thing. Delay the sine wave a quarter of a cycle and you produce the cosine wave, though the attack has been time shifted. Mathematically, if someone said "shift the phase of this sine wave 10 degrees" you would simply time delay or advance the wave 1/36 of a cycle and use your original time points for attack and release to truncate the wave. That is, by definition, what you are doing when you manipulate the phase of a sine wave... time shifting.

Take a look at this neat applet: http://www.udel.edu/idsardi/sinewave/sinewave.html Leave only the blue box checked, and play with the third numerical value - the phase. When you change it by 10, 20, 30 degrees... doesn't that look like simply a time delay of the waveform to you? It is.

I understand what you are getting at, but mathematically it's the same result. I suppose you could go through the trouble of examing a waveform, calculating its frequency, calculating the offset at each sampling point (in the digital sample) to phase shift it, and be done with it. Or, you could slide the waveform in time by the phase shift required, and retain the original time dependent attack and release times to truncate the shifted wave. Identical end result, but probably much less computationally intensive to manipuate the waveform in the time domain instead of offsetting every single sample point.

You can calculate the phase difference between these two and redraw one to match the other without changing the attack and release at all.

Look at that wave applet again, and look at how the attack looks when you have a phase of 0, 90, 180 degrees. There is a very subtle sonic difference in the attack being in a trough, midpoint, or crest of the wave. You yourself said that humans can hear the difference in a polarity change (which I highly doubt in the case of a sustained pitch, but perhapse we can detect differences in attack), which means the attack is still in the midpoint, just that it ascends in level in one case, and descends in the other. Personally, I think such sonic differences will be small in most cases. Perhaps for low frequencies it is more noticeable. What I would be worried about is some cumulative effect of altering the attack characteristic of every single frequency component in the waveform. That might have a noticeable effect... or it might not. Never done a listening test on that one.

And a fourier transformation is a method for solving/manipulating a differential equation...

Well, it's actually a differential equation that transforms a wave from the time domain to the frequency domain. Been a while since I did that though, and don't plan on doing it again if I can avoid it.

[Bigus Dickus]

Originally posted by ebeam
You know, I just thought of a great analogy for phase shift without time delay.

Think about the profile of a corkscrew spinning on a hard, flat surface. The corkscrew's heght doesn't change (time), the thickness (amplitude) and spacing (frequency) don't change throughout the length, just the phase.

Yep, pretty good analogy. But, look at the very tip of the corkscrew (the attack). Doesn't it look different depending on what position you revolve the corkscrew into (phase)?

You can keep the attack at the same place, but it will look different. When I say time shifting the wave, that is really independent of what you do at the beginning or end. You can truncate a wave at any point you choose.

[ebeam]

I guess I didn't understand what you were saying before. A frequency dependent time delay followed by adjustment of the attack and release would be mathematically the same as a phase adustment. Makes sense I guess. I wonder what that would sound like. The shape of the waveform would definitely be changed and by changing the relative phase of different frequencies, you would be altering their interractions and I assume that would change the sound. But, if you could alter the sound so that important frequencies aren't cancelled when combining two signals, a plug-in for this might be useful, or at least interesting.

[Bigus Dickus]

Originally posted by ebeam
I wonder what that would sound like. The shape of the waveform would definitely be changed and by changing the relative phase of different frequencies, you would be altering their interractions and I assume that would change the sound.

Yeah, I'm kinda curious myself now that we've talked about it. I would love to hear a smooth sweep through phase differences to see what the effect is.

However, AFAIK all analog high and low pass filters alter tha phase (dependent on frequency) of the sound passing through them as well. If I remember correctly, low pass inductors lag the sound based on frequency, and high pass capacitors lead the sound.

[ebeam]

Well, EQ has an effect on phase. I wonder if that Waves Linear Phase EQ does something similar to what we are talking about to minimize this effect. It has some serious latency from what I understand.

[RevF]

Looking at it from a simple viewpoint -

If you are just simply switching the phase - you end up with an overall wave change of X1=X-2X (or X1=-X+2X for the -'ve portion of the wave).
ie This is nothing to do with delay (which is cheating by just shifting by a long-wavelength amount and hoping that this cures-all for the other frequencies.)

However this is still only an overall wave view when phase is switched like this, and therefore does not cater exactly for the higher frequencies? (Does this matter? - Can anyone hear this?!)

In theory one would have to have a multi-band phase switcher to do the job properly - which theoretically would have an infinitely small dy/dx bands of frequency to be able to reflect the whole frequency spectrum?
Presumably this would not be possible/practicle to come close to, and if a phase-band approach were attempted, the resulting bands would tend to have some sort of effect on the phase themselves. (Though Waves claim to have minimised this with their LMB compression plug-in)...
ZZZzzzzzzzzzzzz...

Fuel to the fire...
where did I put those matches?...

[participant]

I'd love to see such a plug-in. From what I understand from following this thread, the theory is to try and break a complex wave form into frequency bands and process from there. My question is... do you really have to worry that much about the higher frequencies? I mean, a 1kHz sine wave has a cycle (period) of 1ms. That means that the most you'd have to nudge it in any one direction would be half of that... can daw's even manage that? More importantly... would we even be able to notice it?

You have to shift lower frequencies much further, of course... and this is probably where we'd hear the major difference.

IMO the closest you're going to get in the digital realm is frequency ranges... the digital domain is, of course, made of discrete samples... so applying "continuous" calculus to it is absurd, you'd agree... we couldn't possibly create a plugin that aligned 102Hz, 102.1Hz, 102.2Hz... etc (like this over the whole sound spectrum) perfectly anyway.

Make it 3/4 bands -- adjustable -- with the option of shutting off bands for performance reasons. You could just dial in the "problematic" areas with your ears--for example--center the band around 315 and adjust out "mud"... this would be highly dependent on the mix. Turn on more bands for more processing... simple as that.

So, let's get on Waves/McDSP/Antares/PSPAudioware etc. about this. This could kick ass


Chad

[Hugh Jazz]

Originally posted by participant
... so applying "continuous" calculus to it is absurd, you'd agree... we couldn't possibly create a plugin that aligned 102Hz, 102.1Hz, 102.2Hz... etc (like this over the whole sound spectrum) perfectly anyway.

Make it 3/4 bands -- adjustable -- with the option of shutting off bands for performance reasons. You could just dial in the "problematic" areas with your ears--for example--center the band around 315 and adjust out "mud"... this would be highly dependent on the mix. Turn on more bands for more processing... simple as that.

What about dividing the bands up into octaves or fractions of octaves?

[Bigus Dickus]

Even after you break a signal down into it's constituent frequency components, you can't "align" those components to one another. You can shift them in time, and thus in phase, but there's really no meaning to the term "align" in this context.

If one frequency was 1.13584532189 another frequency, how would you "align" them? It might take hours before you get any type of repetition in their interference pattern.

No, I think you can simply change the sound. Align is a psuedo-term... it's simply what sounds good, and what doesn't.