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Waves vocal rider

Discussion in 'Vocals' started by Nutti, Jan 2, 2014.

  1. Nutti

    Nutti Active Member

    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
     
  2. JohnTodd

    JohnTodd Well-Known Member

    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.
     
  3. pcrecord

    pcrecord Don't you want the best recording like I do ? Well-Known Member

    + 1 on the sidechain use !
     
  4. Nutti

    Nutti Active Member

    Ok, so I bus everything exept vox to one channel and sidechain that to the vocal rider...how much level of send do I add? Just a flat 0db?

    Sent from my GT-I9300 via Tapatalk 2
     
  5. audiokid

    audiokid Staff

    I wouldn't be using anything like this. Not a chance.
     
  6. Paul999

    Paul999 Active Member

    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.
     
  7. audiokid

    audiokid Staff

    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.

    Waves Vocal Rider
    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
     
  8. audiokid

    audiokid Staff

    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.
     
  9. Paul999

    Paul999 Active Member


    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
     
  10. audiokid

    audiokid Staff

    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
     
  11. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    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.
     
  12. Paul999

    Paul999 Active Member

    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?
     
  13. audiokid

    audiokid Staff

    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, $*^t 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...


    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.

    :)
     
  14. Paul999

    Paul999 Active Member

    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.
     
  15. Kurt Foster

    Kurt Foster Distinguished Member

    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.
     
  16. audiokid

    audiokid Staff

    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.
     
  17. Paul999

    Paul999 Active Member

    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.
     
  18. audiokid

    audiokid Staff

    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 $*^t 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. $*^t 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 $*^t 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.
     
  19. Nutti

    Nutti Active Member

    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
     
  20. JohnTodd

    JohnTodd Well-Known Member

    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.
     

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