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#31
2nd November 2005
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I agree in a way; I don't know about the tubes breaking, but after spending years layering, panning, etc. I'm digging the sound of one or two guitar tracks.

I've been liking the sound of the first Bad Co. record; maybe that has something to do with it!

But lots of layering does seem to destroy some of the quality of what you hear with a single track.

I agree.
#32
3rd November 2005
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16 ms of unprocessed delay is a great starting point for guitars.
#33
3rd November 2005
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The other thing you can do is modulate one of the delays (the old chous thing...). Using a depth of 30% of your delay difference and a modulation frequency of 0.3 to 0.6 Hz won't be very noticeable in stereo and should clean up the mono phasyness pretty well.



-tINY

#34
3rd November 2005
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Quote:
Originally Posted by tINY


The other thing you can do is modulate one of the delays (the old chous thing...). Using a depth of 30% of your delay difference and a modulation frequency of 0.3 to 0.6 Hz won't be very noticeable in stereo and should clean up the mono phasyness pretty well.



-tINY

could you explain how to do this? i have no idea.
#35
3rd November 2005
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Use a chorus plugin.
#36
3rd November 2005
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Quote:
Originally Posted by AlexLakis
Use a chorus plugin.
i will have to try that
#37
3rd November 2005
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Quote:
Originally Posted by aetucker1
i recorded that clip that i posted. it wasn't meant to be an example of what i want, it was meant to be an example of tube breakup. since i recorded it, i know that it is one guitar, one take, delayed 6ms and 12ms. i will try a wider delay i guess, but at the delay time of around 30ms the tracks get muddy and the definition is no longer there.
Wow, 6 and 12 ms on that track? Seems longer to me, I can hear a slight slap back. Huh, maybe my ears are better (or worse) than I think they are....



Not a bad sounding clip by the way. thumbsup

Pinksail says 16 ms of delay to start but usually that is too short for me. As he says that is a good starting place but I tend to end up higher when all is said and done. Actually I don't tend to do much in the way of delays on gtr tracks at all, but that is just how I work.

I still don't have a clue about the "tubes breaking" thing, just sounds like a guitar amp cranked up to me. Because I don't get what you are talking about I don't understand why a short delay on the source is effecting it. *shrug*

I will say this, the more you fine tune the delay the more you can get the comb filter to work with the sound instead of against it. If 35ms is getting muddy you might find that 33 ms will cancel the mud that you are hearing. Maybe 30ms will or 36ms so you really have to play with it to get it working. Maybe split the difference and start there, say 25ms or so then play with it to see if you are getting any help from the comb filtering.

I guess I am saying there is no one answer here and as much as I hate to repeat it, use your ears and make subtle adjustments.

Hope this helps a bit.
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#38
3rd November 2005
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Quote:
Originally Posted by not_so_new
Pinksail says 16 ms of delay to start but usually that is too short for me. As he says that is a good starting place but I tend to end up higher when all is said and done. Actually I don't tend to do much in the way of delays on gtr tracks at all, but that is just how I work.
Damn, you just called me Pinksail. Ouch!!!!

Anyway, I was just offering the poster a point of reference with regards to the 16ms settings.

Not a blanket statement by any means.
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#39
3rd November 2005
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Quote:
Originally Posted by picksail
Damn, you just called me Pinksail. Ouch!!!!

Anyway, I was just offering the poster a point of reference with regards to the 16ms settings.

Not a blanket statement by any means.
LOFL...

Sorry man, typo... honest..... really...

Yes I agree not a blanket statement on my part either. Sucks to say but ya gadda use your ears on this one, there is no right or wrong way to go except to say the "right" way is the one that sounds best and the "wrong" way is the one you don't like as much.....

#40
3rd November 2005
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Quote:
Originally Posted by not_so_new

Yes I agree not a blanket statement on my part either. Sucks to say but ya gadda use your ears on this one, there is no right or wrong way to go except to say the "right" way is the one that sounds best and the "wrong" way is the one you don't like as much.....
I think you guys are missing one aspect which i think the poster is missing.

When using an outboard delay you have to factor the built in delay from the conversions(both A/D and D/A).

This delay factors into the actual ms you choose and which unit of choice.

That's why a lot of suggestions are so close and also differ so much.
#41
3rd November 2005
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Quote:
Originally Posted by thethrillfactor
I think you guys are missing one aspect which i think the poster is missing.

When using an outboard delay you have to factor the built in delay from the conversions(both A/D and D/A).

This delay factors into the actual ms you choose and which unit of choice.

That's why a lot of suggestions are so close and also differ so much.

Actually I thought I mentioned that? Let me see.....

Sure as sh*t I edited it out (I have been trying to get my posts shorter ). I wrote a whole paragraph about latency and such but I scraped it because I was not sure the poster was ITB or OFTB....

Yes, this would make a big difference aetucker1. If you are doing this with a stand alone hardware unit then you must factor in the convertor latency. Hell even in the box if you are not using a DAW with plug in latency compensation you are going to get somthing different than you think you are.

If you are in the box you might want to just slip the tracks and find the best place to work from but.. you know, YMMV.
#42
3rd November 2005
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Quote:
Originally Posted by not_so_new
Actually I thought I mentioned that? Let me see.....

Sure as sh*t I edited it out (I have been trying to get my posts shorter ). I wrote a whole paragraph about latency and such but I scraped it because I was not sure the poster was ITB or OFTB....
Its the way i learned which delays work on what.

And which don't and i shouldn't even bother with.
#43
4th November 2005
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well, i am in the box. I have been messing with it a lot and I just dont quite feel like it is the way it should be. Maybe it is just that it is in mono and it just doesn't sound the same. It doesn't necessarily sound wavy, it just sounds weak and not like one signal. It sounds a thin and not precise. I have been messing around with 30ms and lower, but none of it really sounds like i want it. also, does anyone have any good suggestions for a free chorus plugin that will do the modulation deal? Thanks again. also, could you post something that sounds right in mono that has heavy distorted guitar...like open notes strummed. that would be a huge help to me as maybe i just dont know what sounds right.

allen
#44
4th November 2005
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Have you thought about stereo micing? or M/S micing technique? Maybe that is the solution to your problem.
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#45
14th March 2006
Old 14th March 2006
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The Haas Effect

Does anybody know where I can read the original papers on this subject? I really want to get a strong understanding of the science behind the theory but all I can really find are generalities and second hand info.
#46
29th August 2006
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Haas effect!

Bruce will you please give us your word on the haas effect and using effects like stereo spread by panning the same mono sound left and right and pitching 1 up and the other down.

IM curious if you ever do this, when u do it and what you use for the application.


Thank you very much

Bryan Tyson
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#47
30th August 2006
Old 30th August 2006
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May I first caution you!!!!!

Quote:
Originally Posted by vaesion View Post
Bruce will you please give us your word on the haas effect and using effects like stereo spread by panning the same mono sound left and right and pitching 1 up and the other down.

IM curious if you ever do this, when u do it and what you use for the application.

Thank you very much

Bryan Tyson
Bryan.....

Good topic!

HOWEVER.... I see a defiinite tendency here in folks trying too much to come up with something unique with the Nutz And Boltz of our craft and far too little effort is placed in the uniqueness of our own imagination in music recording!!!

To continue .......... - The Haas Effect

The Haas effect can be used to overcome directional masking. Haas says that, in general, echoes occurring within approximately 40ms of the direct sound become fused with the direct sound. We say that the echo becomes "one" with the direct sound, and only a loudness enhancement occurs.

A very important corollary to the Haas effect says that fusion (and loudness enhancement) will occur even if the closely-timed echo comes from a different direction than the original source. However, the brain will continue to recognize (binaurally) the location of the original sound as the proper direction of the source. The Haas effect allows nearby echoes (up to approximately 40ms delay, typically 30ms) to enhance an original sound without confusing its directionality. We can take advantage of the Haas effect to naturally and effectively convert an existing 2-channel recording to a 4-channel or surround medium. When remixing, place a discrete delay in the surround speakers to enhance and extract the original ambience from a previously recorded source! No artificial reverberator is needed if there is sufficient reverberation in the original source. Here's how it works:

Because of the Haas effect, the ear fuses the delayed with the original sound, and still perceives the direct sound as coming from the front speakers. But this does not apply to ambience--ambience will be spread, diffused between the location of the original sound and the delay (in the surround speakers). Thus, the Haas effect only works for correlated material; uncorrelated material (such as natural reverberation) is extracted, enhanced, and spreaddirectionally. Dolby laboratories calls this effect "the magic surround," for they discovered that natural reverberation was extracted to the rear speakers when a delay was applied to them. Dolby also uses an L minus R matrix to further enhance the separation. The wider the bandwidth of the surround system and the more diffuse its character, the more effective the psychoacoustic extraction of ambience to the surround speakers.

Of course there's more to the Haas effect than this simple explanation. To become proficient in using Haas in mixing, study the original papers on the various fusion effects at different delay and amplitude ratios.

May I first caution you!!!!! Something like The Haas Effect, IN NO WAY surpasses what you can add to your music mixing and recording, by developing your "SONIC PERSONALITY"!!! It should be understood, but don't think that it's a big deal in music recording!!! The Haas Effect is available to essentially everyone on the planet!! Your own "SONIC PERSONALITY" is YOURS AND YOURS ALONE!!!

Haas' Relationship to Natural Environments

We may say that the shorter echoes which occur in a natural environment (from nearby wall and floor) are correlated with the original sound, asthey have a direct relationship. The longer reverberation is uncorrelated; it is what we call the ambience of a room. Most dead recording studios havelittle or no ambient field, and the deadest studios have only a few perceptible early reflections to support and enhance the original sound.

In a good stereo recording, the early correlated room reflections are captured with their correct placement; they support the original sound, help us locate the sound source as to distance and do not interfere with left-right orientation. The later uncorrelated reflections, which we call reverberation, naturally contribute to the perception of distance, but because they are uncorrelated with the original source the reverberation does not help us locate the original source in space. This fact explains why the multitrack mixing engineer discovers that adding artificial reverberation to a dry, single-miked instrument may deteriorate the sense of location of that instrument. If the recording engineer uses stereophonic miking techniques and a liver room instead, capturing early reflections on two tracks of the multitrack, the remix engineer will need less artificial reverberation and what littlehe adds can be done convincingly.

Using Frequency Response to Simulate Depth

Another contributor to the sense of distance in a natural acoustic environment is the absorption qualities of air. As the distance from a sound source increases, the apparent high frequency response is reduced. This provides another tool which the recording engineer can use to simulate distance,as our ears have been trained to associate distance with high-frequency rolloff.

An interesting experiment is to alter a treble control while playing back a good orchestral recording.

Notice how the apparent front-to-back depth of the orchestra changes considerably as you manipulate the high frequencies.

AWWWW.... Big Deal!!!!

Bruce Swedien



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#48
6th September 2006
Old 6th September 2006
  #48
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Chi-Lite reflection affection

Quote:
Originally Posted by Bruce Swedien View Post
Bryan.....

We may say that the shorter echoes which occur in a natural environment (from nearby wall and floor) are correlated with the original sound, asthey have a direct relationship. The longer reverberation is uncorrelated; it is what we call the ambience of a room. Most dead recording studios havelittle or no ambient field, and the deadest studios have only a few perceptible early reflections to support and enhance the original sound.

In a good stereo recording, the early correlated room reflections are captured with their correct placement; they support the original sound, help us locate the sound source as to distance and do not interfere with left-right orientation.

Dear Bruce,

mmhh, bit like a brain-bender.

You state, in order to keep the first reflections, you use to locate the microphones away from the singer or the instrument in order to keep as much directional information as possible.

Here is, how I understand it:

Let’s say, I’m in the middle of a square room with the dimensions 34 Metres (~100 feet) x 34 Metres x 16 Metres (~50 feet) high.

If I clap my hands, I’m able to detect my relative spatial position as in the middle because the sound energy of the first reflections arrives to my ears at the same out of either direction within 100 milliseconds, right?

If I move to the side of the room, the ratios of the back-bouncing wavefronts will vary in relation to my position. If I for instance move 8,5 Meters (25 feet) to the side and clap again, the first reflection bouncing back from one sidewall arrives in 150 ms, from the other in 50 ms.
If I was blindfolded I could with a little conditioning determine my position in the room by listening to the first reflection and reflective patterns. This is easily possible in a cubic shaped environment.

Recording studios are often built in trapezoid shape, like Westlake, on order to avoid shatter-echoes and standing waves. Even more, the walls are not plain, but sloped to act as ‘wavebreakers’ in order to get a more diffuse soundfield. To my understanding, you can’t really speak about correlated reflections at all, here.

Never the less you always refer to the studio as one of your preferred locations. So, where does a “live” studio end and a “dead” studio begin? (b.t.w.: I’ve got another post running about studio design)

If I resume your recording techniques; when you capture audio while using tube traps, you even reduce the spatial information of the recording-room.
So, bottom-line, as far as I understand, there is only very few directional information from to catch in for the little poor microphones in recording environments like Westlake.

On another picture, I saw you recording an orchestra (it was about those DECCA-tree, you were using). The room seems cubic shaped, like Abbey Road, for instance. So is that statement of you only true for or in certain acoustical environments?

I’ve been listening to the Chi-Lites “Oh Girl”, where the mouth organ and the lead vocal appear quite distant; the reflections (panned extreme left-right) seem to have a significant delay to the direct signals which appear both in the center. Did you delay the reverb (the famous 125 ms) for the reason to keep up the early reflections of the recording environment, that I assume was built not in trapezodial shape?

Could you explain?

Very appreciated: Boris

#49
3rd January 2007
Old 3rd January 2007
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Haas effect!

Hi - Do you use short delays (Hass effect) on electric guitars?
How you do you guys do it?

Thanks

Booda
#50
3rd January 2007
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Quote:
Originally Posted by Booda View Post
Hi - Do you use short delays (Hass effect) on electric guitars?
How you do you guys do it?

Thanks

Booda
I do.
I pan them and play around with the delay times.

bstrgrds
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#51
3rd January 2007
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Try this:
Depending on the flexibility of your plugins, use either cloned tracks or routing, set up three channels of the same mono guitar track.

Pan these hard left, centre, and hard right. Delay the hard left channel 11ms, and the hard right track 13ms.

It will sound fat and wide, with no discernable delay, because these delays are in that Haas effect area (below 30ms) where they fuse with the direct sound and are not percieved as echos. But they are perceived as direction queues, so your brain thinks it is getting a wide stereo spread.

Play with it for a while, and the ear candy effect starts to become an articificial plastic sound, and the novelty wears off.

Mono your 2bus, to check mono compatibility. Hear how the phase effect dulls the sound. In small doses, with DI signals, the sound might benefit from this phasey dulling effect to take the edge of. It's close to the desirable effect of the early reflections from room sound - but not quite good enough. Try different delay settings. Try different panning settings. Try just one delay, and pan the dry channel hard left and the wet channel hard right.

The delayed sounds can benefit from modulation or pitch shifting. The can also benefit from eq'ing, compressing, distorting - anything to make the signals different assists with the stereo effect.

These are the basic building blocks of stereo chorus or reverb effects - there are many variations on a similar theme. It's the old Beatles ADT revisited, sort of.

It's usually better and more organic sounding to track new parts, to get some subtle variations. But sometimes this artifical sound is perfect for the job.
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#52
3rd January 2007
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I have merged a bunch of threads together so go back and read it all from page 1 if you are very interested in this topic.
#53
3rd January 2007
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#54
3rd January 2007
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Compression is for kids!!!!

Quote:
Originally Posted by gwailoh View Post
This is OT re Haas effect, but if you want to achieve a more striking depth dimension you might experiment with less or even no compression on the stereo mix bus. If your mix bus compression is raising quieter signals it'll have the effect of moving distant instruments forward in the mix. That is, it'll "compress" your depth dimension as well as your dynamics. Try it and see what you think.
WOW!!!!!

YEA!!!!

You are absolutely right! Compression is for kids!!!!

Guess who.....


#55
3rd January 2007
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I am a bit computer challenged.....

Quote:
Originally Posted by BoW View Post
Dear Bruce,

mmhh, bit like a brain-bender.

You state, in order to keep the first reflections, you use to locate the microphones away from the singer or the instrument in order to keep as much directional information as possible.

Here is, how I understand it:

Let’s say, I’m in the middle of a square room with the dimensions 34 Metres (~100 feet) x 34 Metres x 16 Metres (~50 feet) high.

If I clap my hands, I’m able to detect my relative spatial position as in the middle because the sound energy of the first reflections arrives to my ears at the same out of either direction within 100 milliseconds, right?

If I move to the side of the room, the ratios of the back-bouncing wavefronts will vary in relation to my position. If I for instance move 8,5 Meters (25 feet) to the side and clap again, the first reflection bouncing back from one sidewall arrives in 150 ms, from the other in 50 ms.
If I was blindfolded I could with a little conditioning determine my position in the room by listening to the first reflection and reflective patterns. This is easily possible in a cubic shaped environment.

Recording studios are often built in trapezoid shape, like Westlake, on order to avoid shatter-echoes and standing waves. Even more, the walls are not plain, but sloped to act as ‘wavebreakers’ in order to get a more diffuse soundfield. To my understanding, you can’t really speak about correlated reflections at all, here.

Never the less you always refer to the studio as one of your preferred locations. So, where does a “live” studio end and a “dead” studio begin? (b.t.w.: I’ve got another post running about studio design)

If I resume your recording techniques; when you capture audio while using tube traps, you even reduce the spatial information of the recording-room.
So, bottom-line, as far as I understand, there is only very few directional information from to catch in for the little poor microphones in recording environments like Westlake.

On another picture, I saw you recording an orchestra (it was about those DECCA-tree, you were using). The room seems cubic shaped, like Abbey Road, for instance. So is that statement of you only true for or in certain acoustical environments?

I’ve been listening to the Chi-Lites “Oh Girl”, where the mouth organ and the lead vocal appear quite distant; the reflections (panned extreme left-right) seem to have a significant delay to the direct signals which appear both in the center. Did you delay the reverb (the famous 125 ms) for the reason to keep up the early reflections of the recording environment, that I assume was built not in trapezodial shape?

Could you explain?

Very appreciated: Boris

Boris......

I have some photos of the Chi-Lites sessions that I did for "Oh Girl".

PM me your email and I'll send you those photos. You post them for me here and we'll talk and then all the Slutskies can participate. I am a bit computer challenged so It would be great to have some help with getting the photos up here.

By seeing these photos I can show you the "Dry" part of the recording.

Bruce Swedien


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3rd January 2007
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im just getting more and more into this trick for depth. I have been using a stereo delay hard panned synced to 1/8 on 1 side and 1/16 on the other. I send vocals and synths to it...sometimes some hats..whatever works.

It appears that I need to get more into timing out the MS though....
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#57
3rd January 2007
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I use this trick once in a while, on keys, or if I was only able to get one guitar take (because of time limitations, etc.) I usually will pan the original to one side, then delay the other side by 20ms to start (in the box, no latency.) Then listen in mono and adjust the delay time until it sounds good. It's going to sound a little phasey and blurry no matter what, but you can typically find a spot where it still sounds good. Then go back to stereo and check to see that you're getting the effect you want.

I agree that changing the eq, adding some distortion, etc are all helpful. Also, if you're lucky enough to have a DI of the guitar, reamping to a competely different setup for the other side gets a nice fat, but still very tight sound. Or reamping to the same amp setup, but with the gain rolled back a touch, maybe a different mic, etc.

The haas effect is also relevant with regards to drums or any multimiking technique, of course. It refers to the psychological effect of short delays (in opposite ears.) Read about Comb-filtering if you want to know more about the phase things happening, and why collapsing short delays to mono causes problems...

http://www.moultonlabs.com/more/abou...y_reversal/P0/

Good article, simple yet informative!
#58
3rd January 2007
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Quote:
Originally Posted by John Suitcase View Post

I agree that changing the eq, adding some distortion, etc are all helpful. Also, if you're lucky enough to have a DI of the guitar, reamping to a competely different setup for the other side gets a nice fat, but still very tight sound. Or reamping to the same amp setup, but with the gain rolled back a touch, maybe a different mic, etc.



http://www.moultonlabs.com/more/abou...y_reversal/P0/

Good article, simple yet informative!

I have not listened, but it could possibly be the Chevelle record is what you say. One signal two amps. I would bet on that more than the use of delay, that is and was a trendy thing to do.

www.bluethumbproductions.com
#59
15th September 2012
Old 15th September 2012
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Think this thread deserves a bump, a lot of useful information in here. Hope it helps someone else. Definitely have to re - read this and glean as much info as I can from it.
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