x
anonymous 11 October 2001

WAHHHHHH HOOOOOOOOO!

Fukin AMAZING !

First mix done with em, I was running about 25 of them...

Qualities:

"Direct" - there aint no bogus 'veil' across it, this ain't no 'this-will-have-to-do' sound..

"Big" - big sound, great mid control - WITH BALLS -we are talking 'improvement" not 'sound destroying' boost.

"GML like' - surgical but smooth at the same time

"screw the graphic man, I am f**king HEARING it"!

"top end finally useable" - man oh man, super slick, but jumping out at you.

Vital step for the PT comunity, a Sony compressor will complete the package.

Honestly - this LIFTS the gloom of waiting for the new PT hard & soft ware "to get better sound'.

Light at the end of the tunell!

:b:

Topic Tags

Comments

Guest Fri, 10/12/2001 - 04:38

Going in

World class mic's & Di
vintage outboard classics
Apogee AD8000 Special Edition

Monitor of Pro Tools stereo out
Via outputs of AD8000 SE

Whole digital side of studio clocked by Rosendahl Nanosyncs

This week I am just mixing and the Apogee is out on hire and I borrowed a ADAT Bridge i/o (as a work around I can listen to the analog output of my Cranesong Hedd which is in use across the mix bus to add a little phatness if i dont want to listen to the ADAT Bridge..).

I mix 'all in PT'

:)

anonymous Tue, 10/16/2001 - 15:42

I've bought the Oxford Plug-in, too. And all I can say is: FUCKING HELL!!

This puts ProTools on to a total new level. This is the first time you can seriously consider mixing something on that damn thing!

Before the Sony plug-in I used all the others, McDSP, Waves, BF, you name it. I tweaked and tweaked for ages.... and never really arrived.

With the Oxford it's like back in my analog mixing desk days. You'll get an instant result.

My mixes are deeper, wider. And the vocals jump out of the speaker. You activate the high shelf and you not only get 'Air' you also get 'body'

And that's even WITHOUT the GML option!

In January they'll release the compressor. God bless us!

The Oxford plugin will not make everything sound good. But given the right music and engineer, it certainly will.

Steve Rhodes

Guest Thu, 10/18/2001 - 15:43

RTAS planned, as well as other platforms..

I should have a hand to a beta version of the Sony TDM compressor plug in next month. I will report if I am permitted!

Mixing is a fukkin blast now...

The DSP usage is quite heavy, especially if you wish to have the low & hi
>>> filters as well. Here is the spec:
>>>
>>> For MIX Farm cards: EQ & Filters 2 per DSP chip, EQ (no filters)4 per DSP
>>> chip, Filters (only) 3 per DSP chip
>>>
>>> For DSP Farm cards (Merle PCI cards): EQ & Filters NONE (bummer!), EQ (no
>>> filters)1 per DSP chip, Filters (only) 1 per DSP chip
>>>
>>> It will not run at all on Nubus DSP Farm cards.

Audio suite all the other 'toy' plug ins to free up DSP and MAKE ROOM for the real deal!!!!

We have a winner!

Guest Wed, 10/24/2001 - 17:08

Hmmmm!

While other folks have raved about this plug in or that plug in I have remained unimpressed regarding almost all of them as - average, and un special, and frankly a drag to have to use on my audio.

The Sony plug in is a breath of fresh air.

Net chum David Frangioni, (Aerosmiths chief audio computer tech & midi/beats programmer + engineer & producer in his own right), claims it to sound - just like their Sony Oxford console EQ.

Anyhow for those 'stuck at a certain level' with PT audio quality..

Using:

A high quality ext word clock device (ArdsyncII, Nanosyncs, Lucid etc etc)
High quality converters, (Cranesong Hedd, Cranesong Spyder, Apogee Rosetta & AD8000 & PSX100, Lucid 24/96 etc)
a Fatso
and the Sony plug in

Will find a marked improvement..And this might tide you over till the new PT system in spring / summer 2,002.

IMHO

anonymous Thu, 10/25/2001 - 12:08

I'm curious to know if you all think that the Sony Oxford EQ sounds SO brilliant compared to the other Pro Toolsplug-insbecause the standard of the competition in Pro Tools is too low or is it that it really stands up in the wider context of the best digital and analog EQ's?

Thanks
Renie

You nailed my exact same question prior to buying the Sony EQ. We all have a pile of plugins... Many are great for this or that, but nothing is brillant all the time...

With so many bright engineers designing algorithms and writing code (Colin McDowell, Waves, Metric, Digi, even the "message board battered" Eric and all the engineers behind the scenes) and so many years of development on the current platform and DSP type, can anyone really raise the bar for a digital eq???

We all go through the experience of incremental improvements with new plugins and learning the right places to use them, but nothing seems to raise the bar...

Then my thought moved to this - - the Oxford is a console in it's own class. The Oxford architecture has the luxery of (many) custom chips and lots of DSP power. The algorithm design target for the Oxford console is sound quality as the first priority and the hardware was designed around the algorithms to ensure optium sound quality.

So my assumption is that Sony's approach for a Protools plugin was to keep as much of the original Oxford algorithm intact and the main effort was mainly a re-coding process to get their (already well known as great eq [and comp]) algorithm to fit into the Protools DSP with the least ammount of sacrifcie in sound quality....

This approach would be different then most (if not all) of the existing plugins which had a starting of algorithm design and coding only for the Protools DSP. Thus, the difference between starting points could mean something different in the outcome....

I don't think it's a situation that the others are bad, I think it's more that the Sony had a different starting point to the code and algorithm and a deeper foundation of knowledge on digital eqs, while the others are either trying to model and simulate a well known analog eq for a specific purpose (like BF's Pultec), or having a different foundation in their knowledge and design approach to digital eqs... There is certainly a huge difference in analog eqs (and pres, and comps, etc.) with only a couple of caps and resistor changes to a well knon and copied Neve, RCA, etc, circuit, it makes sense the same is true with code...

So, I said what the hell and bought the Sony plugin... well I found in the first few minutes of use, that it saved the snare and guitar tracks on a couple of songs that I was getting to the point of having to re-record, because I couldn't get any plug-in to help them... Now that I've used them for 4 weeks, I still think they're great and they have raised the bar on what protools can do, but also keep in mind there's no single fix...

IMO, probably not the right place for this, but I need to rant a bit on the issues with the protools mix, which I think has less to do with protools and more with so many other factors, like:

#1 Ability of the Engineer is everything... anyone with money can buy good / excellent tools today at very low prices, but good engineers only come with experience, ears, knowlege, experimentation, open minded-ness, etc. Tools are great, but a craftsman can do wonders with any tools, and the novice will usually get crap even with the best tools...

#2 Quality of the tracks getting into protools means so much... mics, pres, eqs, comps, converters make so much difference... crap-in always equals crap-out, but I think so many people expect Protools to fix their bad tracks...

#3 Less is more - It's too easy in protools to put too many plugins on all the tracks. I think we all went nuts when we started with protools doing more then what is good or needed for the "Mix"... I think having too much editing / mixing power and options has caused Protool users to start working a mix first on the track level (ie, stuffing the usual assortment of plugins, and spending way too much time tweaking individual tracks trying to get the perfect track), rather then starting at the mix level and only touching a track if it needs something...

Too many plugins give a processed character to the sound, phase problems (not just DSP delay, but phase problems as what the plugin algorithm does to the signal) starts to stack up on multi mic tracks, and gain blocks can easily get screwed-up quickly, the mix goes away.... Less always works out better for me..

#3 Gain blocks / stages are everyting in the analog world (when to hit it hard, or don't hit hard, etc.), and I think we are still learning what works in the digital world, and what works in our digital architectures (which vary from digal platform design to desig in the same way analog designs vary). There are so many posts and "expert" debates on the subject of digial mixing math and theory, but for me, it has come down to keeping mind of basic rules and always doing what sounds good. For me, that means that I worry much less about getting every last bit of resolution and full scale, and worry so much more about the peaks that I don't see on a meter and getting more headroom then what I used to strive for in analog...

#4 The old tube vs. solid state wars of the 70's and 80's are now the analog versus digital wars. The way energy is transfer and stored (stored on tape, but also stored on the way to tape in devices like tubes or silicon and caps, etc.) is so much different in the way analog works then the way a signal is converted and stored as a number. We are brain washed that the only good sound is what we know and love... I've played guitar for 35 years and I will only play old tube amps (in fact for me mostly old Fenders), why, because I love the way energy is stored and released in a tube design and the way the fender circuit works, because my ear and brain is totally brain washed that this *is the only* good guitar sound. I've tried and owned everything... The new world of amp simulators like the POD, Line 6 amps, Fender Cyber Twin etc. work for some people and certainly can be perfect for some songs and style, but they're still a model/simulation of a tube amp trying to achieve the real tube amp sound in my brain...

So my point is that people who mix on Neve or Focusrite or SSL or ___ consoles in the past will have that sound in the brain and it's so hard to find anything that makes it right... But for people like me that grew out of prosumer boards like Tascam and then Mackie and now DAW mixers, we get excited to have a DAW which is a huge step up from our past mixer, while the Neve / SSL crowd get bummed trying to get a DAW mixer to sound good.

Guest Thu, 10/25/2001 - 14:58

Great post!

"the Neve / SSL crowd get bummed trying to get a DAW mixer to sound good. "

Well that's me BTW... I call it - "tearing my hair out'. Or - "It's driving me nutz"!!!!

Lot's of us, including Fletcher (if I may be so bold) are appoaching mixing in DAW's with the positive attitue, - I'm gonna lick this sucker!

:)

anonymous Tue, 10/30/2001 - 18:37

Buick, PERFECT POST

Jules,
Can you give my a instances per mix farm as opposed to a per chip. (which raises the question, which type of chip?)

I have a mix 3 system and need to know how many I can get up per farm before I would buy it.
Cant afford another mix farm AND the sony plug!

ed

anonymous Wed, 10/31/2001 - 01:17

I came to pro tools from logic audio and a mackie mixer etc. It's great! For me it's a major improvment and I love it. When I read posts saying how crap it sounds I think two things...

1. Lets not get back to the digital analog debate....We'll be here all day angrily agreeing with each other.

2. The price difference between a mix plus system a load of plugs and a procontrol, is not even in the same ball park as an ssl console etc etc etc.

I, as a big fan of pro tools dont have any problems with the way it sounds. If others are used to the cream of the desk crop, maybe they will. BUT That doesn't change the fact that I am happy with the sound. In short, it's always a good thing to re-evaluate the way you listen. If someone can hear a certain something, identify it so that you are aware of it....but if you like something, someone elses observations about it are just that...THEIR observations.

Jack

Bob Olhsson Wed, 10/31/2001 - 04:26

I pretty much agree with Buick all the way but would like to comment on the fourth point since I started out working in tube studios.

The thing that made them great was not the "tube sound" that solid state's apologists invented as their means of defending new solid state products that offered manufacturers unprecedented profit margins.

In the 1950s, every stage of a professional tube console typically clipped at around +35 into a 150 Ohm load. (a lot of common outboard gear did not have this kind of dynamic range, just the consoles.) Mike preamps had fixed gain and occasionally there were some 10 or 20 dB. mike pads in the patch bay. If you measure the dynamic range between noise and 3 percent distortion as opposed to 1 percent distortion, a tube console could have as much as 20 dB. more dynamic range. On top of that you were operating internally at levels well above thermal noise, RFI and crosstalk.

This meant the sweet spot was immense and it hardly mattered where you trimmed the gain in the chain to keep the meters from pinning. It was not uncommon to leave a console set up and for everybody to just walk in and start recording without doing any sound checks at all outside of maybe making sure all the mikes were working before the musicians arrived. Most studios had one or two channels of limiting, a couple Pultecs and maybe a graphic equalizer. (I remember being wowed by four 1176s being available at Wally Heider's in 1971!)

I've never used a solid state product that offered this window of "ease." We evolved into an unimaginable number of bells and whistles combined with unprecedented amounts of studio time being used for the typical recording project so studio businesses didn't feel much pain. Low-end consoles and mixers were even worse. The move from analog recording to 16 bit digital closed off another major portion of the "ease" window.

Potentially 24 bit recording could finally turn the tide of the shrinking sweet spot IF we have people creating applications who really understand the importance of gain structure and headroom. I shudder every time I read about analog "modeling" and "warmth" when the real issue all along has been internal dynamic range and headroom.

anonymous Thu, 11/01/2001 - 09:55

Potentially 24 bit recording could finally turn the tide of the shrinking sweet spot IF we have people creating applications who really understand the importance of gain structure and headroom. I shudder every time I read about analog "modeling" and "warmth" when the real issue all along has been internal dynamic range and headroom.

Aren't we already there with this digital/DAW stuff? At least with 32-bit floating point processing, and it's 1500db of internal dynamic range, we should be pretty close, right?

I was exploring this the other day with my system (not Pro Tools). I put a 16-bit, 0dbfs tone at unity gain across 192 channels, lowered the master fader about 43db, summed them together to 16-bit (with dither on the mix output), and looked at the result. Virtually identical to the source - no distortion artifacts, just a slight increase in the noise floor.

Does this tell me anything meaningful about the system under test, or am I barking up the wrong tree?

BTW, Sony is sending out promotional emails that reference this thread, and others like it. I guess they're pretty happy with Julian's asessment of the Oxford eq plug. :)

Guest Tue, 11/06/2001 - 05:25

RE GML

From Sony..

"The Q is reversed as the 8200 controls work the
other way round to the R3. Basically the bandpass sections of the GML8200 are exactly the same as the Oxford in type1 selection except that the freq ranges are different and crucially the HF section control can go up to 26KHz. The shelves ARE completely different from the Oxford in that they are much much gentler - as you can see on the graph."

26k!!!!!!

:)

anonymous Tue, 12/04/2001 - 04:17

Originally posted by Greg Malcangi:
>

Bob are you sure about this? Surely if you've got a 44.1k file there is nothing above 22kHz. Doubling the sample rate surely can't introduce frequencies above 22kHz that weren't in the original file?

Greg

Doubling the sampling rate is used for reducing midband distortion, and spreading the errors over a wider bandwidth. Result is cleaner than processing @ lower sample rates.

Peace,

Zooot.

x

User login