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ITB vs. console summing test Consoles
Old 30th April 2018
  #91
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Silvertone's Avatar
I love ITB mixing. I think you can make fine records that way provided everything is captured correctly.

For myself, I’ll be summing through this console, these EQ’s , these limiters and printing to a 1957 3 track recorder. I don’t think any form of ITB will create this sound... ever. I can almost guarantee it. lol

Check out the Instagram link below to see the limiters. Couldn’t copy the photos.

Highland Dynamics on Instagram: “BG2 “plus” in BG23 frame. . . . . . . #tubecompressor #tuberecordingstudio #tubeaudio #recordingconsole #recordingdesk #recordingstudio…”
Attached Thumbnails
ITB vs. console summing test-99667677-ece4-4e4b-ab28-8767c60578be.jpg   ITB vs. console summing test-64cbfebc-e941-46b7-9cb9-9d9c733abd51.jpg   ITB vs. console summing test-7e814c17-f84d-4138-b51a-ac34c9c61fde.jpg   ITB vs. console summing test-705f5c6c-87fe-40a9-bdf0-763cf4df8955.jpg  
Old 30th April 2018
  #92
Gosh they look TASTEY
Old 30th April 2018
  #93
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Silvertone's Avatar
JJ, I applaud your efforts here.

I think post like this make many think it has to be one way or the other. Of course this is art so however it is created is correct.

I worked in every way possible to record, still do, still love them all.

I would miss tape if it went away... just like I would miss real photos. Both mediums alter the art...
Old 30th April 2018
  #94
Here is a great example of what the SSL summing sounds like compared to in the box. Like the Dangerous, its not a huge difference but the differences are there and the SSL has a more urgency attack on the transients and it has a more open top end.
YouTube
Old 2nd May 2018
  #95
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jjblair's Avatar
Quote:
Originally Posted by Silvertone View Post
I love ITB mixing. I think you can make fine records that way provided everything is captured correctly.

For myself, I’ll be summing through this console, these EQ’s , these limiters and printing to a 1957 3 track recorder. I don’t think any form of ITB will create this sound... ever. I can almost guarantee it. lol
[/url]
Yeah. That's gonna have a sound for sure!
Old 2nd May 2018
  #96
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bambamboom's Avatar
Quote:
Originally Posted by Jason rocks View Post
Here is a great example of what the SSL summing sounds like compared to DOING NOTHING AT ALL.
YouTube
There, fixed that for you.

You can easily accomplish what the summing is doing in that video entirely ITB.
Old 2nd May 2018
  #97
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Quote:
Originally Posted by bambamboom View Post
You can easily accomplish what the summing is doing in that video entirely ITB.
If it is so easily accomplished why not share how?
Old 2nd May 2018
  #98
Gear Nut
 
cdruzeta's Avatar
In the original AB example I can hear a clear difference in the guitar.
Listen to the tails of the vibrato. I don't know which is which but I can hear that one more clearly rings out over the mix.
Old 3rd May 2018
  #99
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bambamboom's Avatar
Quote:
Originally Posted by johnnyc View Post
If it is so easily accomplished why not share how?
I already have on multiple occasions in different threads. Look up one of my posts.

There are also youtube videos about this.
Old 3rd May 2018
  #100
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Quote:
Originally Posted by bambamboom View Post
I already have on multiple occasions in different threads. Look up one of my posts.

There are also youtube videos about this.
I remember you mentioning it multiple times but never much details, was always "PM me".

The last couple weeks I have been working on simplifying my setup to the bare essentials for when recall is a dominant factor. I tried various console emulations, read online, watched videos. In the end it never sounded as good as summing feeding a mix bus compressor.

So I'm genuinely curious. If it really is easily accomplished then it should be easily explained. If it's already been explained could you link to the thread? I couldn't find it in search.
Old 3rd May 2018
  #101
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Tangent for Jules / Mods to think about adding to this site.

Regardless of this particular thread as the same issue pops up in many, how about adding some file upload space for comparison threads where members can post examples in a proper wave file. Maybe limit it to only a wave extension and stress posters also put a few seconds of 1k tone at the start of each example for easy level matching. You can set time limits the examples will be up so I don't see a huge amount of space needed. Good idea that solves a comon problem here?
Old 4th May 2018
  #102
I think the SSl video is very helpful in demonstrating how it's external summing changes the sound.
Old 4th May 2018
  #103
this feels like a good ole' gearslutz thread with some classic contributors

Thanks for the test audio JJ. A quick listen for me suggests one of these examples (the second in the first A/B) has more low mid energy and centre focus. It sounds louder though so that could be coming into play, perhaps that is some subtle OTB saturation, but perhaps its the different outboard in the ITB chain...it is tough to draw a conclusion as I know the ATR can have quite a low bump on it...Given the panning I'd probably still guess tho that the second example (in the first A/B) is the outboard summed...

I think this test is probably more a 'summing test with my gear how I might use it' by JJ Blair as others have pointed out. However still interesting to have a chat about it esp if posted by experienced 'slutz...
Old 7th May 2018
  #104
Gear Addict
 

Quote:
Originally Posted by jjblair View Post
Mix was broken out of the Apogee Symphony onto 29 channels of the MTA 980 with Inward Connections discrete class A stereo bus, run through some Western Electric 111C transformers -> NTI EQ3 -> Dramastic Obsidian -> Apogee Symphony.

ITB has ATR 102 on the master bus -> Apogee Symphony -> 111C transformers -> NTI EQ3 -> Obsidian -> Manley Vari-Mu -> Apogee Symphony
By the way, do you use Vari-Mu in linked mode or channels separated?
Old 7th May 2018
  #105
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jjblair's Avatar
Quote:
Originally Posted by jdtrbn View Post
By the way, do you use Vari-Mu in linked mode or channels separated?
Stereo linked.
Old 31st May 2018
  #106
Gear Head
 

To those of you who have experimented with sampling frequencies: Is there a sweet spot? Most people seem to track and mix in 96 kHz.
Do plugins benefit from 192kHz?
Old 1st June 2018
  #107
Quote:
Originally Posted by shahstlz View Post
To those of you who have experimented with sampling frequencies: Is there a sweet spot? Most people seem to track and mix in 96 kHz.
Do plugins benefit from 192kHz?
I would say most people tend to track at 48k, closely followed by 44.1.

Higher sample rates are rare in “popular music”. Not unheard of, but not day to day.
Old 1st June 2018
  #108
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Quote:
Originally Posted by psycho_monkey View Post
I would say most people tend to track at 48k, closely followed by 44.1.

Higher sample rates are rare in “popular music”. Not unheard of, but not day to day.
I used to track at 88.2. About 10 years ago, someone here said it speeds up and reduces any errors when converting to 44.1 ...if you do this in the box. What do you think about this concept?

Some of my best recordings were done at 16/44 through my trusty old AD8000 and Rosetta.
Old 1st June 2018
  #109
Quote:
Originally Posted by godlesshorror View Post
I used to track at 88.2. About 10 years ago, someone here said it speeds up and reduces any errors when converting to 44.1 ...if you do this in the box. What do you think about this concept?

Some of my best recordings were done at 16/44 through my trusty old AD8000 and Rosetta.
This is the whole "converting to 44.1 is easier from 88.2 than it is from 96, because it's just half".

This is kind of putting human characteristics onto computers. Computers are good at hard maths!

And of course - if you're mastering OTB it's irrelevant.

Without more details it's difficult to express any more opinion than that!
Old 1st June 2018
  #110
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Quote:
Originally Posted by psycho_monkey View Post
This is the whole "converting to 44.1 is easier from 88.2 than it is from 96, because it's just half".

This is kind of putting human characteristics onto computers. Computers are good at hard maths!

And of course - if you're mastering OTB it's irrelevant.

Without more details it's difficult to express any more opinion than that!
Q. Why is 88.2kHz the best sample rate for recording?
Recording
Published July 2012
I have read that the optimum sample rate to record at is 88.2kHz. The reasons include simple integer-ratio sample-rate conversion, avoiding the phase shifts and ringing of anti-alias filtering at 20kHz, and less data to move about compared to 176.4kHz. Is there any truth in these assertions?


Via SOS web site

SOS Technical Editor Hugh Robjohns replies: These claims are partially true! Let's start with the simple integer-ratio sample-rate conversion issue.

https://www.soundonsound.com/sound-a...rate-recording
Old 1st June 2018
  #111
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This is from Dan Lavry. I think whoever mentioned the conversion to 44.1 being the issue was obviously wrong for the most part. However, according to Dan Lavry 88.2 would sound better than 96k due to accuracy.

Quote:

I see a lot of comments here.

First, it is true that the front end of a modern AD (called the modulator) is running at very high rates (64-1024 faster then 44.1KHz), but it does so for very few bits (typically 1-5 bits). The very fast but few bits data is then converted to the lower rates at many bits (a circuit called the decimator) which is the converter output data rate, thus the sample rate. So one should not confuse localized sample rates of a circuit with the rate of the data used for storage or transmission or DAW processing… the sample rate.

Regarding of the impact of sample rate on accuracy:

We all know that sampling way too slow (say at 1KHZ) is not a good idea. Why? It will only allow frequencies of 500Hz or lower (at best). Sampling that low for audio is simply ridicules.

But we also know that sampling audio at say 1GHz is ridicules. Why? Because at 1GHz we can only get very few bits of accuracy, so the noise and distortions will be really bad. The data size will be huge, and for no good reason at all (we are talking about audio, not 1GHz oscilloscope or telecome gear...).

Therefore, there is a concept we need to adopt – the concept of an OPTIMAL sample rate. Clearly it is above 500Hz, and bellow 1GH. So what is the optimum rate?

The optimum rate for conversion is driven by the application. For weighing scales, the process is slow (a second or more) but we need a lot of accuracy. For video, we need some few MHz bandwidth, but lucky for us, the eye is a lot less accurate then the ear, so having fewer bits (then audio) is OK. For medical applications, it depends on the specific cases (CAT scan, MRI and so on, all have their requirements)…

Generally, if you look at converters, you find out that the faster rates yield less accuracy, the slower rates yield higher accuracy. Why is it so? Well, this is not the only case where doing things faster is a tradeoff for accuracy, and taking your time often enables more precision.

Say you want to charge a cap. If you “take your time”, the final charge will be closer to the intended charging voltage. If you do not wait long enough, it will not charge fully. So you try and reduce the cap size, and now you have other problems show up (beyong the scope of this post)…

Say you want to have an OPamp track an input signal accurately. A circuit designer knows that the best accuracy is at low frequencies, down to DC, and at some point, going to higher frequencies, we lose accuracy. The devices (transistors) inside the amplifier “loose steam” at higher frequencies. Yes you can find real fast transistors, but then you trade off accuracy in different ways.

The charging cap and OPamp are just the first 2 “element” in the AD circuit. There are a lot of caps charging and a lot of OPamps inside an AD… It would take too long to get into the details why speed accuracy is always a tradeoff. Such tradeoffs do exist in electronics and other areas. In electronics one can trade off speed vs. power, size vs. temerature (heat) and so on. It is true that technology is moving forwards, so the tradeoffs today are different then those of say 10 years ago. Please take my comments about tradeoffs as correct for a given time in the history of technology. Lets not compare the speed of today’s gear with that of 40 years ago… Lets talk about the tradeoffs that exist today, or 10 years ago, or at any specific point in time.

For a while, I was interested in writing a whole paper about the technical reasons for speed accuracy tradeoff. But then, I needed to buy a new Audio Precision test system for the production final testing. Now, these guys there at Audio Precision are makers of the finest audio measurement gear, and we are not talking about inexpensive gear. They have a converter based system called ATS2, and you can buy it to accommodate unto a little over 100KHz (for 44-96KHz sampling) at very nice accuracy, or you can buy it with an option to extend to around 200KHz (for 192KHz gear). The point is: if you want to do 192KHz the measurement system is limited to 16 bits! When the best test gear maker has to cut down on precision significantly, for going from 96 to 192KHz, you know that there is speed accuracy tradeoff. Again, the gear I am talking about far from inexpensive. At that level, they do not trade off converter quality for lower price. So when I saw that “proof” that speed costs accuracy, I sort of lost interest in writing a more detailed description as to why speed and accuracy pull in different directions. I can now say
"I rest my case"...

We are talking about audio, so we need to cover the hearing range. What is it? It is a little higher then 20KHz (for some people), so the theory suggests that we can sample at a little higher then 40KHz (twice the bandwidth). But theory assumes that we can “do theoretical things”, in this case, we would need brick wall decimation filters, going from passing audio fully intact at say 20Khz, to blocking it completely at say 20.000001KHz. We can not do such things. We end up needing some margins to “bridge the gap” between theory and good practice.

In my view, that margin should be up to the design engineers. The ear should tells us what we can hear and where the limits are, and the designer gets to find how much margin is sufficient to accomodate the ear. Too often, mastering and recording people, or even gear salesmen, step over thier boundries into the design area, which is a sad fact responsible to the false notion that 192KHz sampling will be better, which is false.

Nowdays there are a number of good designers and ear people that find 60-70KHz sample rate to be the optimal rate for the ear. It is fast enough to include what we can hear, yet slow enough to do it pretty accurately. Faster rate means less accuracy, with some unwanted side effects – increased data size, need for more powerful DSP compute engine, and there is no up side to going faster. Going slower gets the designer “squeezed” at the 20KHz range, which we need to include for high quality audio.

It took a few years to have it turned around, but many of those that “jumped on the 192KHz band wagon baloney”, are coming around to saying that 60-70KHz is optimal. Well, there is no such standard, but 88.2-96KHz is not that far from the optimum. It is slightly faster then I would like, but still acceptable to me.

It is possible that 1-2KHz will be more accurate when using 44.1KHz sample rate. In fact, one can do a great job for 1-2KHz with an 5KHz sampling rate. But the converter designer can not “go there”. We usually need to look at the whole audio range. 44.1KHz can be a somewhat tight squeeze, especially when we keep “piling” attenuation on that 20KHz range – most mics have 3dB loss at around 20KHz, then there is the AD with 3dB at 20K, then the speaker, the processor… Pretty soon the accumulated impact is such that there is not much 20KHz left… Moving the sampling a little higher (be it 60Kh, 88.2 or 96K) takes some of the “offenders” out of the picture (all you need is a few KHz extra and the problem is gone). At my age (62) it makes less of a difference, but a young person with great ears can be impacted.

Regards
Dan Lavry
Lavry Engineering
Old 1st June 2018
  #112
Quote:
Originally Posted by godlesshorror View Post
Q. Why is 88.2kHz the best sample rate for recording?
Recording
Published July 2012
I have read that the optimum sample rate to record at is 88.2kHz. The reasons include simple integer-ratio sample-rate conversion, avoiding the phase shifts and ringing of anti-alias filtering at 20kHz, and less data to move about compared to 176.4kHz. Is there any truth in these assertions?


Via SOS web site

SOS Technical Editor Hugh Robjohns replies: These claims are partially true! Let's start with the simple integer-ratio sample-rate conversion issue.

https://www.soundonsound.com/sound-a...rate-recording
"Simple ratios were important in the days of 'synchronous' sample-rate conversion, but that technology went the way of the dodo a long time ago."

and

"There is no measurable difference in performance between using simple integer ratios or complex ones."

As I said...good references though!
Old 1st June 2018
  #113
Quote:
Originally Posted by godlesshorror View Post
This is from Dan Lavry. I think whoever mentioned the conversion to 44.1 being the issue was obviously wrong for the most part. However, according to Dan Lavry 88.2 would sound better than 96k due to accuracy.
I think you're attaching something to that quote that doesn't exist.

The gist of it is that 60kHz is ideal, and so the next stop on the train is 88.2/96k.

The quote doesn't really express an opinion. I see where you're getting that from, but he doesn't imply there's a significant difference in 88.2 vs 96k.
Old 1st June 2018
  #114
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Guest
I’m neither here nor there on this.
Just posting these for the sake of information and discussion.

You seemed knowledgeable and I thought you might elaborate.
Old 2nd June 2018
  #115
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Optimal sample rate would in part depend on the converter design. I tested this years ago with the Lynx Aurora and from best to worst it went 88.2k, 96k, 48k, 44.1k. Objective measured performance was best at 88.2k.

Integer sample conversion is still actually a thing depending on the implementation. High quality is CPU intensive which becomes an issue with things like real time sample libraries.

As a practical matter CDs are not really a thing anymore so using 48k and it's multiples makes more sense these days imo.
Old 2nd June 2018
  #116
Gear Nut
 

Interesting comparison. Once it gets going, it's difficult to hear when it changes, but I think I like whatever it starts with a little better. I'm guessing it starts with the desk and then starts flipping back and forth. Whatever it starts with seems a little more layed back sounding. Then when it changes it seems to clear up and pop a bit more.

But the difference is subtle. Hope I'm not wrong!

Also, I'm wondering if the comparisons would be more noticeable on say an SSL or a Neve desk between the two. Possibly the MTA is a cleaner sounding desk where the difference isn't as noticeable. Still I hate working on computers, and watching lcd screens all day long. I HATE using a mouse.

Thanks for the test man! Pretty cool.
Old 2nd June 2018
  #117
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Higher sample rate relate to less lag assuming the CPU and transfer rates can keep up.
Old 2nd June 2018
  #118
Quote:
Originally Posted by elegentdrum View Post
Higher sample rate relate to less lag assuming the CPU and transfer rates can keep up.
“Lag” - please elaborate? I don’t feel I’ve experienced this at any sample rate.
Old 2nd June 2018
  #119
Gear Head
Quote:
Originally Posted by psycho_monkey View Post
“Lag” - please elaborate? I don’t feel I’ve experienced this at any sample rate.
I think he means buffersize induced latency. A buffer size of 128 samples will have half the latency in milliseconds on 88khz compared to 44khz. It's a strange way of framing the matter though, because the buffer size is freely assignable and in order to minimize latency it's good practice to set it as low as your system can handle without losing stability. And if your system can handle 88khz at 128 samples, it should also be able to handle 44khz at 64 samples, both resulting in the same latency.
Old 2nd June 2018
  #120
Gear Head
 

Quote:
Originally Posted by psycho_monkey View Post
This is the whole "converting to 44.1 is easier from 88.2 than it is from 96, because it's just half".

This is kind of putting human characteristics onto computers. Computers are good at hard maths!

And of course - if you're mastering OTB it's irrelevant.

Without more details it's difficult to express any more opinion than that!
Interesting information here. Why I brought this up in this thread: Besides an ideal sampe rate for AD conversion while tracking, it's also about mixing ITB:

I would have presumed that the higher the sample rate the more accurate all processing like plugins and summing. So shouldn't a session with massive plugin use sound better at let's say 88,2 than 44,1?
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