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Waves vocal rider

Hi!

So I did my first mix using the vocal rider but I still feel I need to compress vocals before hitting the vocal rider. The vocals where very dynamic so I set the vocal rider to fast and max sensitivity without a compressor but it just sounded as a mess with huge peaks up and down in volume. I tried the soft mode but that just kept the peaks longer, I also tried playing with sensitivity but never got it without compressing vocals first to catch the peaks.

When I bought this plugin I thought it would be the end of compressing vocals and manual volume automation but at this point it doesn't seem so. Am I doing it wrong or do I just need to have a serious talk with the vocalist and get him to work the mic better? I might have the vocal rider at the wrong place in the chain? I have it last in the chain now, perhaps it needs to be ontop?

Sent from my GT-I9300 via Tapatalk 2

Comments

JohnTodd Thu, 01/02/2014 - 10:36

I use it all the time. Some people here disagree with me using it, but I'm not a top-rate engineer so I use it to 'fix' things I shoulda' fixed myself.

You will still need to use a compressor.

Here's the chain I find works best on vocals:

1. EQ: HPF set at about 200Hz. Cut everything below 200Hz to get the mud out. The exact frequency will vary depending on circumstance.

2. Gate: I set my gate to gently roll off 1/4 second after threshold, and usually from 5-15 ms of attack. YMMV.

3. Waves Vocal Rider. By the time the signal gets to the Rider, the mud has been cut out and the "room" has been gated out, so it won't respond to those things. I usually set mine to fast and max sensitivity, but it depends.

Also, don't forget to activate the sidechain (check the manual), and then send some things into that sidechain. Half of what the Rider does is based on what comes down the sidechain. Without it, you're getting half the usage.

And remember to adjust those little arrows and stuff to the left of that big fader. That determines the range of volume the Rider is allowed to operate in, and sets it's "resting" position. Very important, too.

Hope this helps!
-Johntodd

PS: Get the vocalist to work the mic properly is ALWAYS going to make a better recording!

PPS: Compress after the Rider.

Paul999 Thu, 01/02/2014 - 21:01

audiokid, post: 409351 wrote: I wouldn't be using anything like this. Not a chance.

LOL. It is a useful plug actually. I tend to use it between a couple of compressors. I'll compress a little to get attacks like I want. Next I'll use waves vocal rider. I do not let it use massive range. Then I'll finish. Expecting vocal rider to do the heavy lifting is unrealistic.

I use use it about 15% of the time. Anytime I feel I'm having to get heavy handed with compression or starting to over process.

audiokid Thu, 01/02/2014 - 21:58

I guess out of sheer ignorance or lack of hearing it, I figure, if a track is this bad, do it over. But if it saves a bunch of labour and can kiss something enough without leaving residue or an automated feel to it, right on.

I had to look more into this now.

[[url=http://[/URL]="http://www.soundons…"]Waves Vocal Rider[/]="http://www.soundons…"]Waves Vocal Rider[/]

I found Vocal Rider fairly intuitive in use and the results sounded very natural, allowing me to be more discriminating in the use of any compression subsequently added. To achieve the same level of consistency through manually writing automation data would be extremely time-consuming and probably still wouldn’t achieve such precise results, so I don’t think I’m going too far in suggesting that this could be one of the most significant new ideas in plug-ins since automatic pitch correction. And, like most good ideas, it seems entirely obvious when viewed with the benefit of hindsight.

Ya know me. I'm running for the hills on most things today lol. I miss the sound of mistakes done well.

I've never noticed one plug-in to date that didn't leave something behind that was worth what it did well. So, every time I use something, especially on the center Vox, snare, kick and bass, I always think about what I am steeling away from the entire mix, not just the track, but the entire mix. Or, what I'm doing to the center to rub it with the sides.
Side chaining comps are very useful so it sounds like this is somewhere between.
I never knock something before you try it and I definitely admit, I've never tried it. Maybe I should.

I'd love to here a before and after

audiokid Thu, 01/02/2014 - 22:12

John, I think you are way to critical with yourself and quite possibly, reaching for a style at times that causes you to be less satisfied with your natural talent and direction. I think we all do this at some point in our career . Unfortunately, we often don't find this out until later in our lives.

I'm waiting for that track you send me some day. :) You got 20 year to go before I forget what I just said. lol.

Paul999 Fri, 01/03/2014 - 11:26

audiokid, post: 409364 wrote:

I've never noticed one plug-in to date that didn't leave something behind that was worth what it did well. So, every time I use something, especially on the center Vox, snare, kick and bass, I always think about what I am steeling away from the entire mix, not just the track, but the entire mix. Or, what I'm doing to the center to rub it with the sides.
Side chaining comps are very useful so it sounds like this is somewhere between.
I never knock something before you try it and I definitely admit, I've never tried it. Maybe I should.

I'd love to here a before and after

I just have to play devils advocate for a minute and have some philosophical banter with you on this:).

I never hear plugs steal anything. That is the beauty of them. They leave nothing behind(with obvious exeptions). For years I made pilot error decisions with plug ins. They are so transparent that before I knew it I was killing something. Compared to hardware were you can hear it just plugging it in and it generally will take a bit more heavy handed approach. A daw eq or compressor is going to steal nothing unless used badly.

When I hear you caution people about using plugs especially if everything is ITB it sounds like the audio recorded is this delicate flower that might wilt if you look at it wrong. Well recorded audio should be resilient and be able to withstand a s@&t kicking when mixing.

I know know your a very competant engineer and that you and some of the guys here are hearing phase in places I can't imagine. I feel like you guys got sucked into some alternate plane of existence and I don hear what you guys hear either that or it just doesn't impact my brain as being important. I just think a noob reading your post would be afraid to insert a plug after. That would be a set back and a shame.

I'd love to hear a before and after that shows a plug stealing something.

Cheers

audiokid Fri, 01/03/2014 - 13:43

Yikes!

I've been hearing the negative effects plug-ins have since I stared using DAWs for years now. Lots of factors and some could be in my head too. I don't rule anything out.

I hear subtle to quite obvious once something is in the loop. Some plugs are more so than others. Bad code maybe, dunno.
Its also why I prefer Native code over third party but that's just me. Some things I don't need proof on, just hear it. The whole plug-in marketplace just starts looking like a good thing to avoid.
I tend to think the stock stuff tested is all we should need. Its tested to work on that version of that DAW. The extra stuff is why I personally go OTB. But, some plugs like Fab Filter Pro L are on my capture DAW all the time. yet, I still get squeamish just seeing it on lol..
Also, I love object based editing. That's to die for. You can process clips of any length (objects) and glue it, then remove the plug-in from the DAW. All the processing it took is moot. I'd do that over riding or de-essing if i was really serious about the sound.

Other than bad code, (just a guess), buffer speed and CPU usage is effected through plug-in track activation "yet another plug just in case I need it" thus causing latency. This might not appear to be showing an error on your screen but it sure is in the audio! I know we have latency compensation while you round trip, but that's another patch. Then there is automation along with the plug-in(s) on any given track. Everything added is taking a piece out of the original session.

Its insane to think the DAW and your basic clock is not misfiring or dropping a bit here too.
I hear audible change even when plug-in are in the path, flat. Audio is going through them. Or do you think it doesn't until you change a value?

This (philosophical banter) could be what some of us refer to as hearing phase shift when anything is moved or added to a session process. Thus, why external clocking has also been recommended by top level engineers around the world for the last 2 decades..

I dunno Paul, I stopped thinking DAWs were so perfectly pure and transparent years ago.

Recently on GS doing ADC shootouts, I've even heard Pro Tools tracks hold an effect when the client claimed it was disabled. A ghost effect was obvious on the nulled R side of a track. Its was confirmed reverb was off but it was clearly on the right track with no business being there. It left him scratching and others unable to comment.
Some DAW's can even keep a plug-in, in the loop when disabled. If you don't reboot, it can keep hanging. Worse yet, You have to dump the entire session if you want it gone because its somewhere in one of those bits that you saved 1 hour ago and never noticed it until too late! Don't kid yourself. If you haven't noticed this stuff, you aren't listening.

Noted: Since we are all using different systems, everything is subjective so I never say someone isn't experiencing something like what you are accusing me of..

Therefore, I tend to follow the rule, if you don't need it, don't use it. Don't even activate it. In fact, try everything in your power over ever relying on plug-ins claiming to glue something that needs this much automation. Learn to sing or enjoy your sound instead of patching something that will never be performed.
To me, this vox rider effect and most others like this are nonsense. But hey, look how autotune has made people millions.

I've been a sampling, workstation freak for 35 years now so it isn't my "age" talking here, its my ears. I've heard negative effects from computers for years. Nothing's changed about digital. Its about bits, processing, conflicts and bugs. And, too many things happening at once.

Regarding noobs or pro's:
A good place to start any session = less is more, thin out processing whenever you can.

imho

kmetal Fri, 01/03/2014 - 20:36

My aha moment on this was w/ PT 7, i was slowly starting to notice degradation at the studio by putting many instances of the liquid mix on a mix, and kept wondering why it lost depth and fullness, even tho these comps/eqs were vintage modeled units designed to do that very thing.

so i went home, and i booted up my system, just a basic system. laptop cheap interface mackie hr's. i had a raw track i knew very well, and instantiated the digi eq 3 pluggin. listened, disabled it, and didn't hear a difference. when i enabled it, then removed it completely, and the instantiated it again, and that was it. i heard the same thing i did at the studio, albiet subtle, my sound lost dimension, and somehow sounded a bit more grainy.

i'm not saying i have golden ears, and i'm not just echoing the sentiments of people far more experienced than i am. But i do hear it.

i think bob katz put it very well, when he said 'as soon as you alter a digital file in any way, (fade pluggin edit) ect., you create a new digital file, defined by a new binary code.' thats not the verbatim quote, but pretty close.

while i can't claim to hear anomalies, of fades or edits, i hear it in plugins. I mean if you need some 10k air on a vocal track, you have to use what you got, if you need it, and it would probably benefit more than it would degrade.

but to me plug-insaren't the same as a digital board i use where i cannot hear a difference when i bypass or enable, the eq/comp, which is stock on every track. in this system the audio in the daw is basically a tape machine. it's converted thru the board. but these eqs are very transparent, but very musical. extremely useful for surgical stuff. by no means anything special.

i think alot of people got fooled into plugins, forgetting that alot of the pop songs they liked were made from professional samples (808 anyone), sent raw to a mixer like CLA, who has a ton of OB, and an SSL console. oh he uses waves thats all i need.

and even listen to the 'best' or 'current' top 40 mixes they have tons of digi eq, or some other pet favs (i see (vids/mags) alot of Renaissance comp/eq, and a surprising amount of stock avid) from the pro's, they aren't even better than 90's stuff like tori amos, most of the time.

I've been doing alot of research on this type of thing as i've come into like 60k worth of additional equipment down at the studio, bunch of mics pre's comps, couple eqs, and a trident 24 board. (not mine, but i drool, and feel lucky to even be in the same room with all that stuff)

these are some of the justifications i have heard from some interviews w/ top 40 guys, who aren't analog freaks.

time, they get files from all different people in different time zones, and need to sent multiple versions to multiple people, which sounds like a logistical nightmare. Instant recall is crucial. Just as important, is they may work on 3 or more projects in one day, and there is no time to readjust tons of stuff.

also w/ so many systems involved between artists producers, it is not fair to expect that everyone is going to have all the same plugins, or OB. So another good reason to use stock plugs at least until the final mixer. again, time and convenience.

I think sound quality is pushed into 2nd place, w/ the champs being loudness, and deadlines. my world doesn't have this tight of a schedule, nor do i make anywhere near what these guys get paid. if the money was there and my skills were, i'd do the compromise to, if thats what the client needs, thats what they get. the end listener is just used to the sound they are given, and would probably think a mix that didn't sound like this, wasn't 'good'. who am i to judge?

i'm taking all the measures i can to make an what i feel are improvements to what i hear on top 40, or at least not try to fall into the same deadline driven pitfalls. can i do it? i dunno, but i'm not convinced that a cpu should be made to do much more than be act as tape machine(s)/editors.

The other thing is how many more times can you listen to another vocal w/ the soundtoys decapitator? i like don't even wanna use it any more, and i like it. still sneak it on snare tho. wheher its auto tune or whatever 'everybody's got it' pluggin, it just gets sickening when people who don't who use words like soundcard, use the same some dave pensado uses. it's not being elitist, it's sickening that all those top guys are using the same stuff too. it's like you can just buy 'the sound of the week'. and sure the same could be said for hardware, but not every studio would have the same setup at the same time, and evrything sounds a bit different. this is where ITB plug-insmake the sound (subjectively to me) boring, dated, and unoriginal. again not that hardware isn't, it's not like we haven't heard 57 on a stack, or a lexicon box, a million times. but i feel that many ITB pluggs are both technically, and creatively reducing the quality of the final products, in a very general sense.

i think that plug-insare really cool in concept, and i use them (as needed, really needed), and i think w/ the advent of DSP devices that are responsible for nothing but effx, it's going to allow, for a more component based 'everything does it's own job' type digital or hyrbid system.

i'm far from a seasoned vet, but i urge you to try the process that opened my ears, and see if maybe you can hear a difference too. maybe i'm just crazy. i mean i think we've all twisted a knob for a few seconds and either convinced ourselves it made a difference, or just the opposite and wondered whats wrong. either way it took from like the late 1800's to get analog 'right', so i still have faith that digital will not only catch up, but surpass. i mean i'm sure people will always wanna have that particular sound for some reason, but digital is still coming into it's own. and i think plug-inshave not reached they're full potential.

Paul999 Fri, 01/03/2014 - 21:47

So then to add some science into the mix even though I am not a sciency kind of guy here is the experiment I purpose.

I'll put a wave file on two tracks. I'll enable a 3rd party eq plug in not actually working on one track and reverse phase on the other. If you are both correct these files shouldn't null because the results are audible to you.

1. This is will be my experiment tomorrow and I'll take it a step further. I'll put 10 3rd party plugs turned on on one track with none on the other and see if they null.

If they Null I'll go a step further

2. I'll duplicate both tracks 10 times. If these null I'll go one step further.

3. I'll use 10 eqs with a frequency boosted 10 DB's on 10 tracks. On the other tracks that previously had no plugs I'll add 10 eqs doing the reverse eq.

My hypothesis is that these will all null perfectly. I will post my results.

Are there any other ways you would refine this experiment to get to the bottom of this?

audiokid Fri, 01/03/2014 - 23:55

lol,

You are barking up the wrong tree here. I'm so (quality made) analog heavy, completely sold on hybrid and transparent DAW processing, completely blown away over the uncoupled capture systems and what external clocking doesn't do, but reveals.
I cannot say enough about Sequoia/Samplitude and I'm so bent on using less and less third party plug-ins, you will never change the reasons why I pass on my experiences. Agree or not, I'm so excited about what I hear.
If 50% of what I think ever meets popularity, I would be amazed. I am not following the mass DAW movement.

So, you can try and null something like this but your thinking is so far off the map on this one, you will surely end up learning nothing useful. But fly at it.
fwiw, (but not for my benefit), I'd rather see you save a session clean, then process it like you do for a week or two, then, (if at all possible) see if the original waves null with the finished. Is that possible, I dunno. Maybe just listening is enough.
You will surely discover a degree of phase you've injected into your mixes.

Everyone is welcome to keep sharing disbelief, questioning to downright calling someone out like this, its healthy! :) Thats what keeps us all honest and makes a better world.
However, you'll never, and I mean never, convince me that you are remotely accurate on the purity of digital and itb music. It looks good on paper, and mass needs the affordable recording, but it sure the hell isn't remotely stable like you think it is. Especially when things start getting congested.
When code is combined with other code, shit happens and this is most likely why you are searching for answers.

No attack on you, generally speaking.
I'm guessing your system is out of phase and you cannot hear half of what is happening in the digital domain, when it does. And if you think I'm wrong, try demoing a 10M clock and join the fast growing opinions of world class recording and mastering engineers.
If you are doing round trip processing, like most everyone hybrid, your system is out of phase to some degree.
And, if you are out even a bit... , you cannot hear the tiny shifts that count when they appear. Plain and simple fact.

Random glitches are notorious with digital audio processing ( cause and effect through plug-in cross pollination( is that a new term! hehe) .

Maybe you are one of the lucky ones in a million with a perfect DAW system. ;)

Something relevant...

Although latency can make life hard when you’re recording, or playing virtual instruments, it’s normally much less of an issue when mixing: you’re able to set your audio interface’s buffer size to maximum to reduce the burden on your computer’s processor — enabling you to run more plug-ins in the mix — at the expense of increased latency.

Most modern DAW programs now include ADC, or Automatic Delay Compensation, whereby, in essence, the software calculates how much time a plug-in takes to process the audio that passes through it and adds that amount of time to the other channels. Thus even if you’re using a convolution plug-in, or something running on a DSP platform such as UAD, with inherent latency, the computer nudges all the other tracks accordingly. (The notable exceptions here are Pro Tools LE and Pro Tools M-Powered, neither of which feature ADC — you have to calculate and make adjustments manually).

In software-only setups, this usually works flawlessly: the plug-in ‘declares’ its latency to the host software, and the host makes calculations based on that. When mixing using automation, this setup poses no problems: the lag in tweaking some plug-in controls and hearing the results can be a pain if the buffer value is set too high — but it’s still nothing like the show-stopping problem it can be when tracking or trying to play virtual instruments in real time.

“What has all this to do with hybrid systems?”, I hear you ask. Well, although generation loss from A-D/D-A conversion is not such an issue these days, each round trip from the digital to the analog domain adds latency. But all your software knows is that it’s sending audio out and back in: it has no way of knowing automatically how long the round trip takes, and therefore the prospect of automatic delay compensation becomes more challenging. Again, thankfully, most DAW software now includes a means of measuring the time it takes for the audio to make a round trip out of the DAW, through a piece of gear and back. All you need to is set up the ‘hardware plug-in’ (see box earlier in this article) and ‘ping’ a signal through it: your DAW should calculate the latency automatically.

the computer nudges all the other tracks accordingly. Is that not a laugh! I wonder why in phase high end analog is so smooth. Have you ever been completely happy with detent? Its convenient but far from accurate when it comes to dead lock. Is digital detent? Well, welcome to the 10M.

I'm reading the word "calculations" said many ways here.. Its one of thousand of article over the cause and effect of processing. Yet, we trust the DAW is getting it all right between the steps.
If you think your DAW is so special that it is actually keeping an eye on every detail without moving something off a hair, Fly at it.
If you think all plugins are equal or that they do not do something the second they are added into a loop, I'm speechless.

You are correct, your proposed null test should be exactly as you expect.

:)

Paul999 Sat, 01/04/2014 - 09:36

Ah. I am not trying to convince you of my ideas Chris. I am trying to convince me that your hypothesis about plugins is true. I am not trying to dismiss your hybrid belief system. I am trying to prove it one point at a time. My null experiment should assist in this. I'd think you'd be excited about that.

I certainly am not trying to call you a moron. You do have a different philosophical approach then any other engineer I've met. The rational thing to do in such circumstances is to try and prove the validity of the new ideas.

Why would you disengage from the conversation by saying ”you'll never pin me down". You are more then welcome to pin me down on anything I say. I welcome it. I thought you were hungry for knowledge.

You said I'm sure to learn nothing useful with this experiment. Help me design and experiment that is useful. If what your hearing is so obvious it shouldn't be that hard do make a repeatable experiment.

Your response is pretty closed minded and I am pretty disappointed.

Kurt Foster Sat, 01/04/2014 - 10:23

i can't fathom any one thinking that asking a computer to do more and more calculations on a file will not at some point begin to degrade the playback of the file. i have heard this myself.

if you are using outboard gear in the mix the coversion to analog and re sampling back to digital will have an effect as well. now if there's latency (which there always is) the recalculation by the host computer is yet one more task we are asking the cpu to handle and there's more degradation of the playback. it's miniscule but it's there. start adding it all up and you can really hear it. if you can't you're in the wrong business. it's just how it is.

it's not hard to prove it out, i've seen and heard it hundreds of times .... just record a raw basic tracks as best you can and revel in how good it sounds. add more tracks and then try mixing them itb using compression and eq reverbs etc. it never sounds as good as the raw pass or the first basic tracks. it always gets worse. this is because we keep putting more and more straw on the poor camels back. sooner or later it's all the poor beast can handle and it starts performing at a less efficient level.

when we use the DAW for what it does best (recording / playback and editing) and then use analog (with it's infinite processing abilities) to do the mix / effects / playback) we don't run into those issues. there's a reason all be big boys and girls do it this way. it works! it sounds better.

audiokid Sat, 01/04/2014 - 11:00

Paul999, post: 409396 wrote: Ah. I am not trying to convince you of my ideas Chris. I am trying to convince me that your hypothesis about plugins is true. I am not trying to dismiss your hybrid belief system. I am trying to prove it one point at a time. My null experiment should assist in this. I'd think you'd be excited about that.

I certainly am not trying to call you a moron. You do have a different philosophical approach then any other engineer I've met. The rational thing to do in such circumstances is to try and prove the validity of the new ideas.

Why would you disengage from the conversation by saying ”you'll never pin me down". You are more then welcome to pin me down on anything I say. I welcome it. I thought you were hungry for knowledge.

You said I'm sure to learn nothing useful with this experiment. Help me design and experiment that is useful. If what your hearing is so obvious it shouldn't be that hard do make a repeatable experiment.

Your response is pretty closed minded and I am pretty disappointed.

I've edited my post since your reply, I should have closed it off instead. Now see you've responded to it, so, maybe re read it again as I've tried to articulate my thoughts better.

Paul999 Sat, 01/04/2014 - 11:35

Well I did the experiment that doesn't mean anything to you guys but did help me.

I did the experiment in logic with logics basic eq and waves q 10. I turned all the eq points on with no boost or decrease in gain. I used a 44.1 16 bit mastered file.

1. In step one of the experiment with one track with eq and phase reverse and the other with nothing I got a perfect null with logic and the waves eq.

2. I then added 10 instances of each. Again perfect null.

3. I then duplicated the tracks with the waves q10 plug in ten times and the normal track 10 times. Again perfect Null.

So at this point I have 100 eq plugs running, 20 tracks with 16bit/44.1 and 10 gain plugs(only used to reverse phase). A total of 110 plugs and a perfect null.

4. The last step gave me problems. My goal was to have the eqs doing work. Boosting the waves q10 eq 16 db at 2000hz (a very audible frequency) and then decreasing it on the opposing track did not null. I realized I would need to boost it the same on both tracks to get them to null. Duh!

So one waves q10 boosted 16 db on a track phase reversed with another track with the waves eq with the same settings nulled perfectly. I then started adding eqs. After adding 4 eqs to each track they did not null. I figured out that the gain staging was brutal because they were all hitting the red. After properly gain staging the plugs 10 on each side the nulled perfectly. 10 tracks of this nulled perfectly.

So in conclusion LPX can run 200 waves q10 plugs with all points on, boosting 16db at 2000hz plus 10 phase reverse plugins with 20 16/44.1 tracks and perfectly null.

I learned a couple things.
Pay more attention to plug in gain staging. It is wise to always have doubt in yourself.

Now please proceed to tell me how this experiment shows nothing.

audiokid Sat, 01/04/2014 - 12:02

To put it bluntly, lets bust some balls here, there is no other other way.

Start learning more about clocking and you will find the answers Paul. If you are really thirsty for the truth, understand why people think this 10M is helping. That will encourage you to study the cause and effect rather than patching a broken system with a product that can be avoided .

In all my posts, I say this passively. I've discovered something that is what the big boys know or will find out.

If you don't have the process to hear it, this shit is going to go right over your head as well. Some things also cost money but as Bos, Kurt and others have mentioned more than once, a simple console with prior understanding as to why we are doing this is key. It isn't the analog hype here either. Its the ability to avoid the trap as long as you can.
And that is where experience is the clear winner to hearing clear sound and keeping it all the way home. When your are locked in to the perfect phase, you never stop smiling.

Glue is the new word for phase lol.

You cannot produce tight music on some basic DAW doing the round trip without problems. You cannot expect a DAW to think and avoid the detail a human hears between the lines. Shit happens.
You need to get closer between the steps if you want a tighter lock. I'm not saying the computer is at fault. I'm saying, either side of the step isn't close enough when it all starts becoming an automated process. When this starts compounding, its a phase nightmare.

To be honest, I try and not push this dollar factor in peoples faces either. I know most of us can't afford what we do so we compromise. Mass are living with these DAW and once you get the taste for good sound, the next thing we do is follow the blind leaders or the once doing it like us. And let me say this. There are some pretty dated and ignorant leaders compared to the select few that are really knowing whats going on.

This post started a long time back. Look at the guys in here. Some used be members here ranting about the round trip. What a joke.
http://www.gearslutz.com/board/high-end/443209-antelope-audio-10m-atomic-clock-getting-sold-off-owners.html

See more
(Dead Link Removed)
https://www.google.ca/search?q=The+10M+Antelope+&ie=utf-8&oe=utf-8&rls=org.mozilla:en-US:official&client=firefox-a&gws_rd=cr&ei=3F_IUvSYFcfhoAT-lIHwDw#q=The+10M+Antelope&rls=org.mozilla:en-US:official&tbm=vid

I could get specific but ya can't prove this shit on a forum, nor do I really want to. They are correct, the 10M is definitely helping ( I own one too) but, I don't need it anymore because I'm avoiding the shifts before they start. But, if you are part of the broken system like most of these people are, you cannot change someones workflow. Its the blind leading the blind on this and its a business of Avid and clients who are all needing fixing. Its one big mess.

Start reading more into the steps between each byte. That's where the transients live and where Pro Audio really is going, and why a null test isn't going to tell you what you really need to know.
Once you digitize something, its in the system. The key is knowing how to preserve it as long as you can before it reaches the system. If you are all bent on using plugs right in the beginning, and continue to do so until the song is done, man.... is so friken cooked, you cannot return.
Don't take my word on it. Start thinking about the steps clocking is doing ( SRC and bytes and correlation it all has to phase) . We all have our way to the finish line. Some people doing VSTi are pretty much out of this issue. They are working with cooked music already in the system. But, if you are blending real music to digital. This is the stuff you need to understand better.

What Kurt just said is spot on.

Nutti Sat, 01/04/2014 - 12:37

Jeesus...so what I'm doing is pointless? I should quit mixing and just start cooking food instead or just play golf? As I can't and never will be able to afford working with analog gear I might as well give it up? As should everyone else on the planet not strictly using analog gear?

Sent from my GT-I9300 via Tapatalk 2

JohnTodd Sat, 01/04/2014 - 12:43

I'm totally ITB, no analog gear.

BUT, there are 2 things: 1) I'm an amateur, and 2) I believe some day the DAW world will get it right. SOME DAY they will make DAWs/plugins that sound better than analog. But that time isn't here yet.

EDIT:

OOPS, misread your post.

audiokid is essentially correct. Let the DAW do what it does best and let hardware do what it does best. Can't make a silk purse out of a sow's ear.

Kurt Foster Sat, 01/04/2014 - 13:15

it's not the fault of the DAW design. it's simply the indisputable fact that a computers resources have a limit. and no matter how fast a computer is no matter how much ram or HD space it has there will always be a limit. anlog processing is infinite ... no limits.

even a simple no headroom mixer like a Mackie can sum better than a DAW with more dimension and headroom. digital mix bus' suck.

Paul999 Sat, 01/04/2014 - 13:53

I believe that I have absolutely proven that Logic LPX (not top of the line) and my 2 year old imac( best you could buy at the time) is perfectly capable of having zero calculating errors with 210 plugs actively processing with 20 wave files. The output meter showed zero DBS of level and I could hear nothing.

Yet multiple people here claim they can hear just one instance of a plugin being turned on effect their stereo image.

Kurt- I agree that great hardware on a bus sounds great. I am not a fan of mackie summing at all. It should be easy to test summing errors in a Daw.

If summing in a Daw causes errors you could send a mix to an aux bus thus causing it to have its errors. You could then send it to a different bus and phase reverse. If they null there are no errors happening or they both have exactly the same errors?????

I do believe that summing through an SSL sigma or a Neve console would sound great because the hardware sounds great not because digital summing sucks.

Back on topic though all of you guys that hear plug in initialized on a track post the null track and describe your methods so we can make repeatable results. Please show me one plugin or daw that does this.

audiokid Sat, 01/04/2014 - 13:53

Kurt Foster, post: 409409 wrote: it's not the fault of the DAW design. it's simply the indisputable fact that a computers resources have a limit. and no matter how fast a computer is no matter how much ram or HD space it has there will always be a limit. anlog processing is infinite ... no limits.

even a simple no headroom mixer like a Mackie can sum better than a DAW with more dimension and headroom. digital mix bus' suck.

exactly, and from what I am hearing, sum into a capture DAW and a cheapo console will sound even better.

Nutti, post: 409406 wrote: Jeesus...so what I'm doing is pointless? I should quit mixing and just start cooking food instead or just play golf? As I can't and never will be able to afford working with analog gear I might as well give it up? As should everyone else on the planet not strictly using analog gear?

Sent from my GT-I9300 via Tapatalk 2

So, even the modest home studio has better options that I think will sound better than the stupid way we are all going today. The plug-in marketplace is totally gone stupid and its reflecting in all our music.

I'm certainly not going to stop using my DAW. I love mine, but I know its limitations and therefor, keep it where it belongs and do not try and make it do everything in one process.

Big topic, simple solution that doesn't have to cost as much as we think.

audiokid Sat, 01/04/2014 - 13:59

Paul999, post: 409410 wrote: I believe that I have absolutely proven that Logic LPX (not top of the line) and my 2 year old imac( best you could buy at the time) is perfectly capable of having zero calculating errors with 210 plugs actively processing with 20 wave files. The output meter showed zero DBS of level and I could hear nothing.

Yet multiple people here claim they can hear just one instance of a plugin being turned on effect their stereo image.

Kurt- I agree that great hardware on a bus sounds great. I am not a fan of mackie summing at all. It should be easy to test summing errors in a Daw.

If summing in a Daw causes errors you could send a mix to an aux bus thus causing it to have its errors. You could then send it to a different bus and phase reverse. If they null there are no errors happening or they both have exactly the same errors?????

I do believe that summing through an SSL sigma or a Neve console would sound great because the hardware sounds great not because digital summing sucks.

Back on topic though all of you guys that hear plug in initialized on a track post the null track and describe your methods so we can make repeatable results. Please show me one plugin or daw that does this.

I'm not disputing anything that you are describing here Paul. You are completely missing the point I'm on. Why do you think all the guys in that link I just posted claim the 10M is helping them?

Paul999 Sat, 01/04/2014 - 14:09

I approach recording from a business perspective. I can't buy things I can't hear and can't prove. I'll pay big bucks for gear that makes a difference and I certainly have. I do have to make responsible decisions for my business. I spend money on gear that earns its keep. Room treatment, great mics, great pres, good converters and monitors all make sense to me. Things that make my workflow faster like a great control surface full talkback and cue facilities this all makes sense. Great sounding plugins that do a thing also make sense. Avoiding digital without proof seems extreme.

Trusting yourself when we are all capable of grabbing an in active channel of eq adjusting knobs and thinking its sounding better is extreme (don't tell me you guys have never done this). Our ears and our bias is the least trustable information we have.

I find it it arrogant to trust oneself as much as you guys seem to.

Paul999 Sat, 01/04/2014 - 14:12

JohnTodd, post: 409413 wrote: Serious question:

Can I plug the output from my [[url=http://[/URL]="http://www.presonus…"]FP10[/]="http://www.presonus…"]FP10[/] into a nice stereo preamp and then back into the FP10s converters? Or would that cause a feedback loop?

Yes you can. You need to make sure you are routing things correctly in your daw to not cause feedback. Keep your speakers turned down just in case.

Paul999 Sat, 01/04/2014 - 14:21

audiokid, post: 409412 wrote: I'm not disputing anything that you are describing here Paul. You are completely missing the point I'm on. Why do you think all the guys in that link I just posted claim the 10M is helping them?

Good point. I think they are caught in bias as well. I've used several clocks and never ever heard a difference or improvement with any of them. I've used cheap m-audio, presonus, internal Mac and mid level Alesis adat, apogee, and high level, RME, API, waves maxxbcl clocks and could not hear any difference in any of them.

If clocking is so important and DAWs/computers are so messed up I should be able to hear something when I change clocks. I hear nothing. Not even a sideways move.

audiokid Sat, 01/04/2014 - 15:08

Paul999, post: 409416 wrote: Good point. I think they are caught in bias as well. I've used several clocks and never ever heard a difference or improvement with any of them. I've used cheap m-audio, presonus, internal Mac and mid level Alesis adat, apogee, and high level, RME, API, waves maxxbcl clocks and could not hear any difference in any of them.

If clocking is so important and DAWs/computers are so messed up I should be able to hear something when I change clocks. I hear nothing. Not even a sideways move.

Exactly.

audiokid Sat, 01/04/2014 - 15:44

Paul999, post: 409414 wrote:

I find it it arrogant to trust oneself as much as you guys seem to.

Paul999, post: 409416 wrote: Good point. I think they are caught in bias as well. I've used several clocks and never ever heard a difference or improvement with any of them. I've used cheap m-audio, presonus, internal Mac and mid level Alesis adat, apogee, and high level, RME, API, waves maxxbcl clocks and could not hear any difference in any of them.

If clocking is so important and DAWs/computers are so messed up I should be able to hear something when I change clocks. I hear nothing. Not even a sideways move.

I get the continued feeling from you, those who claim to hear and share the cause and effects mentioned are full of themselves. Therefore we should stop BS everyone for the sake of not misleading all the "noobs". To believe you over our own ears and keep the concept of affordable recording alive and on track is not my intention at this point of my career. Nor is giving less than my best in anything I say or do. I'm not here to promote dilution.
Sounds like you are more like us than you realize. Except, we hear something you don't.
Whats ironic , you've tested this stuff and now claim its all BS. Yet, all these guys claim to hear something too. I tend to agree, but for a different reason.

Good on you!

I'm not saying these guys are right or wrong , they are clearly hearing something but the reason they are hearing something isn't ("arrogant to trust oneself"). There is a reason and that reason is what this ongoing discussion is all about.

Anyway, its not my intention to change you or anyone. I feel these things do more harm than good and are a crutch and contributor to the demise of sound and talent. I would love to hear John sing something without it once.

That's all.

Cheers!

Paul999 Sat, 01/04/2014 - 18:40

Yes Chris they are hearing something. So are the pro producers that believe high end cables sound better or tie little bags of stones to their power cables claiming almost word for word the same things these guys are. There are many Hoaxes to be had. The most controversial subjects get the most attention and consequently get the highest price tags associated to them.

I am am not saying this product is a hoax. Mr. Powers says that the mixes are 1 db louder which could be the result of less phase cancelation. That would be cool. As you say in your system with two computers it doesn't help because they are completely isolated. Cool! I am sure your system sounds excellent. Fine. The beliefs you espouse are built on a house of cards. Also fine. That is the human condition. I am attempting to deal with one card. The one where you state plugins take something from the mix by being there. Clearly this is not the case in my experiment unless your claiming that by initializing a plugin the 2-bus was effected(your not claiming this are you?).

Why would you shy away from looking at this one statement? You stated a plug in takes something away by just being there. I challenged setting up an experiment disproving this statement and you agree with my findings. Your rebuttable is the send me looking for fairy dust in clocks because I somehow missed the point.

No the point is You stated a plug in takes something away by just being there. I challenged setting up an experiment disproving this statement and you agree with my findings. Full stop.

audiokid Sat, 01/04/2014 - 20:05

Oh man...

Again, I'll say it and you will push to claim a null test proves something. Are you on some mission here? hehe. I know, you are wanting clarification. smoke

-----------------------------

Regarding the sound of a plug-in. Do you think all plugins are transparent, when sold as being transparent?

Okay, back to ghosts and a long winded reply that I'm about to not even read over for typos. I'm so sick of this topic.

Try it like this.

I believe these issue happen more so when a session is running well into the mix. I know this happens because I've heard this many times over years. I don't believe we can isolate this quite so easy from a clean session. If so, every plug-in would be worth thousand.

When I say, a plug-in takes as much as it gives meant exactly this. If all you do is tweak something, what else did this do? I say, everything has a price. Do you use a de-esser? I will, if I have to but I would much rather find every freq and clip it off on the wave over riding it with an automated process. Why? Because it sounds like ass compared.

Can I prove this to you? Sequoia is super tight. Its a mastering DAW. The code is solid. If it did that, it would be junk and the ME around the world would drop it like a stone.
. I also refuse to instal third party plugs. Its why I paid $3000 for the software. It works and doesn't need anything more.

Does Pro Tools work? If so, why does everyone want third party software? Why are people buying clocks and outboard gear and UAD cards and more interfaces for it? Think about that.

Back to the ghost.

I could with Sonar and definitely with Pro Tools without a doubt years ago. I stopped using that system in 2006 because it sucked. I have heard this on tracks since and also had engineers claim they know this. Its not a new claim. But, I know its subjective to the individual.

Its not a matter of opening a plugin and checking it. I have heard ghosts as far back as the early 80's. When ever there are tracks sequencing , sharing aux and common effects , I've also noticed this to happen more . DAW's are a sequencer/ sampler. You know this right?

This is just a guess.
Code can sit attached in a time line in a way that its like bleed or a trigger or a cue, or an control point. People that have grown up in the digital sampling area with sequencers know it happens and accept it. So we avoid the places that are prone to jitter or dropout. Its why I tend to appreciate quantization over leaving audio or midi real time on either side of the center line of the beat. ( I don't another way to explain that one right now) . But quantization comes with a price which is a dead world. Same dead world digital lives as we keep processing something.

We also know how audio tracks are effected when you make them busier than another, the point of too much going on. When you put a heavier load on a track within a time line , the CP is always calculating it. If there is a plug-in conflicting somewhere and its on a track, with automation, it can create a jolt that is enough to shift something. When you compound these over thousands of processes, alot of things can start happening. I believe this is why I hear phase.
The engineers will often never notice this until well into the session where the song (sequence) sounds swishy. Its a sound that I know right away. Its a subtle thing I suppose, but it effects the hats, cymbals to my ear. I actually heard it on your track . I'm not trying to be rude, just saying it like it is so you understand what I'm talking about. I blame digital processing on this. Its an ugly sounds that is in everything today.

Everyone is effected by this. But, if you are on it from the beginning tracks will end up big and punchy.

I also believe this to be related to why these top level engineers are hearing correction in the DAW when using this super clock. I can't prove it, but I can prove that I don't need it either. I own one. But, I also don't round trip and will never. Its the most problematic thing I've ever heard people do. I cannot even believe people do that.
I believe round trip to be the cause of mass phase issues. Which I believe I hear in your system too.

You use Logic? Well, I know world class engineers who use Logic and they are having clocking issues. How come you don't? Why do you want out board gear and why have you tried all this stuff that doesn't make one bit of difference? Everything I try has a sound , cause and effect. I hear it all. I would never claim someone is full of it when they claim something is audible to them. I know enough about this business that it is more subjective than meets the eye. I am more prone to believe everything sound pretty good, even better when you are in phase. Every mic I have sounds pretty good to me. And that is over 70.

And cable... I tend to agree with you about Vovox vs generic, however, I use power conditioning so my system is isolated from stuff outside the farm. Did it help. **** ya! My rig is so quiet, I can crank it wide open and barely know its on. My converters love it! I cannot rave enough about that one.

Have you ever tried using a capture DAW and avoiding the round trip? I bet not. You should one day.

I'm done.

Long live the Vocal Rider.

Paul999 Sat, 01/04/2014 - 21:51

I'm not going to say you don't hear what you hear. I have been questioning your belief about what is making you hear what you hear. I won't be doing that anymore in this thread. I'm sure your glad for that:)

You are making some claims some of which I have no reason to doubt others which I do.

A. Plugins cause phase shift just by being turned on.
B. Having Sequoia on 2 computers and isolating them with hardware in the middle makes it so you can use an internal clock on each computer without phase shift as an issue.
C. Pro music is riddled with phase issues.
D. All digital manipulation leaves traces that never disappear.

I have no issues with B or C. A and D have been talked about enough:)

What I do have an issue with is the explanations you use to justify WHY you hear what you hear. There is no proof of bad code or otherwise flawed audio engines. At least you've not provided proof of any. Why would a computer and DAW record so pristinely yet fail to add a plugin with such prejudice? Why would it play back an unmanipulated track so flawlessly yet sum with thousands of tiny errors? I have doubts that computers are so selective about when they make errors and when they don't.

I have no doubt you hear what you hear or that you are getting the results you say with computer errors. To me your explanations of WHY sound more like superstition rather then evidence grounded in fact. This doesn't mean you don't hear what you hear. I think you MAY be mixed up as to WHY you hear what you hear.

The more likely cause of phase you hear is not the round trip but the hardware itself. Think about this. Do you think you could get 200 high quality SPL hardware eqs and 2 neos consoles to null as perfectly as I got logic to null today with 200 eqs and 20 tracks of audio. I doubt it. Hardware has much more variation and "error" then digital seems to which would easily account for phase in multi tracked sources such as drums. So if a drum set is round tripped out through hardware, hardware would be a MORE likely culprit for phase issues then software. I am not stating this as being any more fact then your "bad code" imperial fact.

I am not the person to debate this tech stuff with especially computer code as it is not my strong suit. I do not believe you have offered evidence that you are either. You've heard what you've heard. I don't believe you know why. I do believe you've found ways to get around the problem you hear. Cool.

BTW Tomorrow I will mix to an adat HD24 and compare to a round trip ITB mix through my 2-bus and see if I hear a difference and see if they null.

Kurt Foster Sat, 01/04/2014 - 22:28

Paul999, post: 409414 wrote: I approach recording from a business perspective. I can't buy things I can't hear and can't prove. I'll pay big bucks for gear that makes a difference and I certainly have. I do have to make responsible decisions for my business. I spend money on gear that earns its keep. Room treatment, great mics, great pres, good converters and monitors all make sense to me. Things that make my workflow faster like a great control surface full talkback and cue facilities this all makes sense. Great sounding plugins that do a thing also make sense. Avoiding digital without proof seems extreme.

Trusting yourself when we are all capable of grabbing an in active channel of eq adjusting knobs and thinking its sounding better is extreme (don't tell me you guys have never done this). Our ears and our bias is the least trustable information we have.

I find it it arrogant to trust oneself as much as you guys seem to.

let's try to not make this personal ...

i am not advocating any expenditure. i'm not saying go buy this or that. i am only saying that summing analog sounds better than summing digitally. if you think it doesn't or that digital summing sounds better or there is absolutely no difference based on a test you performed on your computer fine. if you trust what a computer shows you over what your ears tell you, good for you. i hear a difference and it's not subtle.

if you don't trust you own ears, you should be flipping burgers, not mixing audio ....

audiokid Sat, 01/04/2014 - 23:39

Paul999, post: 409435 wrote:

The more likely cause of phase you hear is not the round trip but the hardware itself.

Indeed. not to mention.... the time it takes to return back to the session.

The concept that the DAW will correct this latency to exact is also something I doubt is that accurate. But hey, everyone thinks its rocking solid.

So lets put this together...

Why would you want your analog hardware ( any hardware) to remain constant in the first place? Does any of this make sense to you? Not to mention, why would you want to redigitize it two more times before it is even summed with the final mix? You are forcing two extra SR ( ADDA) and time shifting it as well. You get this right? I bet you are using plug-in on that round trip too?

Summery:

You mix ITB, DAC a track OTB to an analog product ( that you expect would add some glue) then return it BACK to the same session in hopes it not only improved the SOUND of that track, but even more ironic, re sampled it back to the original session SR in hope it made a difference.

And now, you are going to tell us that it will null or something.
I'm currious as to why you would want your lush analog track to sound similar to what it was before it left? I mean, you are returning it back to the same session. Do you ever think that the analog is in better shape before it returns back to that same session?

I'm sorry but its no wonder you are having trouble with this.

Consider for a moment the analog loop that you mentioned that routes out of a DAW and straight back in again. This is going to incur a quality loss due to D-A-D conversion with two lots of anti-aliaising filters, and nothing positive to show for it. Because it's in and out of the same DAW, the input and output sampling rates have to be identical and phase-locked, and there is no randomization of sampling instants.

If you take the analog output from one DAW, put it through an analog process that's more than a piece of wire and then re-sample in an independent second DAW, you have uncoupled the first output from the second input. The non-synchronous sampling of the second DAW, even if it's at the same nominal frequency as the first, results in the digital values of the re-captured waveform being different from the ones put out by the first DAW, even with no intentional process such as amplitude change or application of EQ being performed by the analog process. It doesn't matter whether there is a sample-rate reduction between the two DAWs or whether they run at the same nominal frequency, the point is that you get a different set of numbers describing your mix.

There are several things you can get out of this. The first relates to down-sampling (e.g. the 96KHz to 44.1KHz I mentioned in my earlier post). Performing the sample rate conversion (SRC) by re-constructing an analog waveform at 96KHz and re-sampling it at 44.1KHz eliminates the use of digital SRCs, which I have shown in demonstrations introduce subtle but unpleasant high-frequency effects. It also means that the 44.1KHz input sees the high-bandwidth analog input as a natural analog waveform, as though from the output of a pre-amp without having had any of the 44.1KHz anti-aliaising filters applied to it. It will, of course, have been through 96KHz anti-aliaising filters, but the effects they introduce in the high passband are well outside the audio range. Doing a digital summation in a DAW running at 44.1 or 48KHz with many input channels each of which has been through an anti-aliasing filter tends to sum the effects of the filter as well as the audio, and I maintain these effects have an audible component. It was things like the effect of summing multiple channels each of which had been subject to a 44.1KHz filter that I was trying to get away from when talking previously about trying to emulate the quality of the old direct-to-disk recordings.

When considering this hybrid approach, not only is it necessary to have two separate computers or other digital devices (such as HD24XRs) with no sampling clock lock between them, it's also important that there is some external analog processing in the mix path to introduce such things as amplitude change and frequency-dependent phase-shift that emulate the real-world processes that the human ear is naturally comfortable with. In Chris's case, he has built up to having the very highest quality analog processes in the mix path, so achieves the pinnacle of performance. In my case, I use more modest processes such as standard but high-quality mixing desks, and yet achieve a sonic quality using this method that I have not been able to match using a DAW.

audiokid Sun, 01/05/2014 - 00:02

Paul999, post: 409435 wrote:

BTW Tomorrow I will mix to an adat HD24 and compare to a round trip ITB mix through my 2-bus and see if I hear a difference and see if they null.

This doesn't sound right and indeed, lets try and not get personal.
Lets see if we are on the same page here.

Are you saying you will not perform one round trip process and mix the entire session OTB ( all tracks arriving into an analog console the same time) (how many stems?) then sum it OTB and capture this to an HD24 at 44.1

Then you will do (what you normally do) on a single DAW system (including round trip processing / D-A-D conversion within this session ) and sum this version ITB and bounce to 44.1 correct?

If this is the process, you have it closer. Anything less, is a joke.

audiokid Sun, 01/05/2014 - 00:13

If you want to compare you need to do it right.
example of my path steps and SR choices:

1.) > Tracking = (32 AD@ 384/192/96/88.2/48/44.1) > DAW1 . No Round Trip > buses and aux continue to step 2, 3 and 4. (however, 88.2 or 44.1 are my preferred).

2.) > Mixing = 32 DA aux or bus outs> analog console or summing system = (100% analog single path)> insert mixing and 2-bus hardware via (transparent digital patchbays and console inserts, mix using both DAW and console to taste (full analog mix continues) to steps 3 and 4.

3.) > Finishing or Mastering = 2-bus 100% analog flow > mastering matrix > insert mastering hardware via, digital patchbay or Liaison > 2-bus continues uncoupled (full analog mix continues) continues to step 4.

4.) > 2-bus ADC@ 44.1 ( or desired SR) > DAW2 = digital mastering > Dither > CD , DVD, MP3 > online. DONE. Or, simply capture to your ME specs.

4.1) 4 point monitoring via an independent control system on stage4 has advantages. From the very start of a session and anywhere in between, I have full view of every audio step to the finish line. Monitoring any other way is less proficient and less accurate..

This entire process enables you to bypass any part of this process (great for electronic music) without SRC or unnecessary SRC , latency compensation and additional external clocking.
SRC is not necessary from DAW1 to DAW2 as it is uncoupled passing 100% analog flow to the finish line. No external clock is ever needed.

This isn't a competition for me. Remember, you came after me on this one.
Do that tomorrow and compare it to your round trip version. I'd love to hear what you come up with. And I don't mean your ability. Phase between our two systems and overall size we end up with is what I'm talking about.

--------------------------

fwiw,

Back to your quest: To my knowledge the only way to compare/hear the difference in shift and size between each of our methods would be for people to have the same clean tracks and finish them totally separate through their process. But, to hear the full meal deal of what I'm experiencing, this would be a difficult challenge if the original tracks had round trip injection on the stems or tracks provided. I wouldn't want any of that nonsense in something I was truly serious about.

like the comparison of the chef trying to unbake the cake.
Once it's baked, it's baked.

Anyway, its cool you are in search of something. I get that :)
I understand you are spending money and investing in new so I hope this all helps you.

kmetal Sun, 01/05/2014 - 02:50

Yet multiple people here claim they can hear just one instance of a plugin being turned on effect their stereo image.

well i could. cuz apparently pro tools m-powered doesn't even have plugin delay compensation. as far as the liquid mix, my best guess is the quality of the dsp chip design, the buffer size (2048) and the bandwidth of the firewire bus.

i'm not a super tech, so i can't explain other than what i hear, which is similar to the characteristics of something that introduces phase cancellation (thinning, depth collapse), and a buildup of nasty upper mid range frequencies, and grainy sound, which i have heard in badly designed conversion/interfaces, like the digi 002.

again my real point in all this is overuse of plugins. if i had 32 channels of nady eq's comps, and a few berhinger verbs, i would think about what really would benefit from their use, if anything at all. same for plugins. which is where i believe people alot of people miss the boat. the first thing my boss told me was to take the plug-insoff my mix i submitted for my application, i did, and it changed my mixes ever since. they were fuller raw, than hyper processed. i started using buses instead of verbs/delays on individual tracks, and leaving well recorded things alone. but thats just me everybody has their own way.

i'm a big fan of [="http://www.pensadosplace.tv/category/into-the-lair/"]Into The Lair - Pensado's Place[/]="http://www.pensados…"]Into The Lair - Pensado's Place[/] and he uses real top dollar commercial releases for his examples. notice how few plug-inshe uses, and how he employs mostly four buses, and does the bulk of the processing itb, w/ a couple pet OB pieces.

here's another one [[url=http://="http://www.youtube…"]Engineer Makes Rihanna's "Diamonds" Shine Bright - YouTube[/]="http://www.youtube…"]Engineer Makes Rihanna's "Diamonds" Shine Bright - YouTube[/] Rhianna's smash hit "diamonds" well over 80 tracks, notice how few plugins, and the mention that he also uses a few pieces of OB. i'm not making this stuff up.

as far as a null test goes, that still doesnt address anything about the subjective quality to the sound that a pluggin is adding.

imagine a ruler flat group of analog eqs for instance, with a 1k test tone running through it will show flat response at 1k, even if you stacked them 10 times in a row, and it still showed flat at 1k, it doesn't take into account the fact that the noise has stacked, and you can hear it. so even if it's still flat at 1k, when you listen to it you still hear an increase in noise floor.

so back to the bob katz thing, any time you change your digtal audio you change the code. and to me w/ plug-insthat change adds up into the things i've described. if i wanted to be a computer scientist i would've been. but technological tests, and numbers don't always relate to the subjective art. as far as i'm concerned i don't belive their is no way to eliminate the problem at this point, but only to minimize it.

ps- by the time i could afford the trinity/antelope, they'd be on trinity/antelope mk5. i'm interestid this to try and make my investments last longer before upgrade, and possibly maintain value like my amps, and mics, and guitars.

anonymous Sun, 01/05/2014 - 03:48

At the very least, could we all agree that no two plugs are the same when comparing apples to apples? Is the difference in the code? The processing chain? The conversion? The phase? Somewhere else?

There's no doubt in my mind (or in my ears either) that different plug manufacturers sound different from one another. I didn't say necessarily "better", I said "different", although I've heard certain plug ins sound better than others. And, this being the case, could we not also assume that it's also possible that in being different characteristically, that it will certainly effect the outcome sonically?

I agree with Paul that plugs are more "transparent" in terms of coloration in the "classic" sense. There's no way a plug will sound like a real LA2 or an 1176. One is a digital emulation and the other has electronic wires and and maybe even tubes - gear that gets hot - and when you compare the real thing to the emulation of an effect, or, the mathematical processing of to 1's and 0's.... there will be a sonic difference.
So, in that regard, it's correct to say that one is more "transparent" than the other.

I can only say that in my own experience (and ears) that I've found that plug ins treat the sonics differently than analog gear does, and accordingly, the sonics sound different, and often, unpleasant.

Hell, for that matter, I can also personally attest to and say that no two DAW programs are alike. I've used Sonar for years, I know it blindfolded.... yet when I tried Samplitude several months back, I heard a difference. Comparing two exact mix projects (apples to apples, no processing, just raw wav files played back at the exact same levels and pan settings) between Samplitude and Sonar, Samplitude sounded better to me...more defined, less "smeary". Is this a phase thing? Perhaps. It could be several different things, as mentioned above, it could be in the code itself.

Very few have the budget to buy the top notch gear, be it OB boutique or multi-thousand dollar external clocks, so they make the best of the gear that they have with their situation at hand, and, let's face it, I don't think that plug ins are going anywhere.

As Chris mentioned, the best thing to do is to be choosy about what you process.... and, how you process it. To simply throw compressor plugs on every single track, without regard to whether it's actually needed or not, is, I think, a bit foolish, if for no other reason than that you'd be taxing your processor unnecessarily.

New users in particular are more apt to go crazy with the amount of per track processing they use, either because they think that they need to, or, because they are trying to justify their purchase of a library of processors, or, simply because they are ignorant.... how many times have we all answered questions here from new users who are inserting gain reduction on every track, yet their question is always something similar to "so what does a compressor/limiter actually do?"

IMHO, I think it's naive to think that these ITB processors aren't effecting the overall sonics... after all, on one hand, they are supposed to. The problem is that, on the other hand, they are not always effecting the sonics for the better.

IMHO of course.

-d.

JohnTodd Sun, 01/05/2014 - 04:11

All this goes back to what I was saying about my use of the Vocal Rider:

I use it as a crutch. It really does help me a lot because I track in an untreated room and I don't have a lot of skill at this.

So, for me, the Vocal Rider is a blessing. But, understand this, it is correcting deficiencies in my recordings.

If I could get it right when tracking I probably wouldn't need it.

There will come a day when I outgrow it. It's like training wheels on a bike: you need it for a while, then you learn to ride without it.

When I get acoustic treatment for my room, and when I get better mics/preamps, and better engineering skills, my vocal tracks will probably sound better without it.

I never meant to start an argument here. Let's go back to being friends again, OK?

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