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Completely out of my element - looking to learn.

Discussion in 'Recording' started by haus, Jan 8, 2013.

  1. haus

    haus Active Member

    I'm at a loss here...

    10 years ago I had a Dell straight from the factory. When I plugged a guitat into a preamp and ran it into the line in on the MOBO, I was able to record audio and monitor it in real time using the Acid 4 software.

    Fast forward to today. I have Acid Studio 8, and I am looking to record some audio. However, there is significant lag between the preamp and the computer speaker output. You can hear this by turning on the click on the preamp and comparing the headphone output of this unit to the speaker output of the PC. (tick-tock... tick-tock... sounds like 2 metronomes running, not 1.)

    My setup is bare bones and as follows:
    Windows 7 Home Premium 64-bit
    *All updates
    ASUS M4A79XTD EVO Motherboard
    AMD Athlon II X4 630 2.8GHz Socket AM3 95W Quad-Core
    SAPPHIRE 100293DP Radeon HD 5570 1GB 128-bit DDR3 PCI Express
    GeIl EVO One (3x2GB) DDR3 1333
    Main Drive is a SATA 250GB Seagate
    Secondary Drive 1.5TB Western Digital
    OCZ Fatal1ty 550 Watt Power Supply

    I took it to a PC magic-man to "tweak" the BIOS settings. As of 45 minutes ago, I am $65 poorer with the exact same problem I have been trying to fix for months. I am not opposed to building a completely new tower that is specifically built for recording - but I'd love to fix this one if possible. Perhaps the steps we take will lead to a new system. What am I doing wrong?

    (I had an internal soundcard installed (Creative X-fi) but I pulled it, because it gave the exact same results.)

    Thanks so much for any and all help.
  2. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    ok. Latency is what you should search for. You should ask for your money back, if the tweak guy is reputable he will. otherwise you've been had. he didn't fix the problem because he didn't know the answer, which is why he needs to give you the money, or the answer. over here it's free. don't feel bad i got had by a shady mechanic for 1200, non-refunded, he ripped off a high school student. i needed then to pay 400 more to have it properly repaired, at a reputable shop. all he did was "bleed my brake calipers" so my car would drive off the station, by the time i got home (3min drive) same 'ol stuff. what he needed to do was replace the master brake cylinder. i'll tell ya, i'm peaceful mello dude, but, it took everything in my being to not really give this guy some physical problems along w/ my large group of friends who were very willing. by gone's, it takes a real jerk to rip off a high school kid, and just as much as a jerk to rip anyone off. my zen? he went out of business a few years ago so apparently his methods turned off alot of people.

    ok your problem is that your audio goes from guitar converts to numbers in cpu, cpu processes it, then it turns back to the electricity that tells your speakers how to move. analog digital conversion. this causes latency, the delay your hearing while the audio is making the round trip.

    what you are missing is an 'audio interface'. which is essentially a higher (or at least adjustable) performance soundcard. ok so you picked one. (try tascam, motu, RME for some baslines among the levels, but they all do the same basic thing)

    there are two options from this point. Some interface/software offers what's called 'zero latency monitoring' or more commonly 'direct monitoring'. this is basically like listening thru an analog mixer, the signal you are playing/hearing never gets converted to the 1's and 0's (binary code), it never makes the round trip thru the computer, just simply thru the in's out's of your hardware (audio interface). you will not hear any effects applied in your software during tracking, only on playback, cuz the sound you are hearing never goes thru the computer software.

    the other option is based on your software/interface's settings and it does make the round trip. this allows you to hear whatever your doing in the software as far as effects. this is when you start to balance your computing power vs latency (delay) time.

    2 things matter here, 'buffer size' and 'sample rate'. you'll find these in your softwares 'audio settings, or audio engine' type menu item. i kinda look as buffer size as like momentum, or a 'head start' in a race. the larger the buffer size the longer the head-start. so with a long head start you may get 3 strums of a chord out, before you hear the first. it's essentially a delay time, which is measured in (milliseconds, ms). so when your tracking you want the delay time (latency) as short as possible. you want the audio back you your ears asap. this takes up the most RAM, and processing, cuz your computer is transmitting as much 'real time' input as possible, as fast as possible. so you want as small as a buffer as you can get w/ out clicks and pops, or other errors. when your tracking.

    higher sample rates have lower latency. i don't have a real solid reason for it other than, cuz the manual and programs say so. but they way i look at it is kicking a few fist size garden stones a few feet vs gravel. there is just more info getting thru the pipline. there's just more detail being processed in real-time. this again is at the expense of RAM/processing.

    generally you gotta pick your poison w/ sample rates. i think there are only a couple of DAWS that support multiple sample rates w/ in a session. i still use 44.1 (cd quality) or 96 (hd quality). depends on what and ahere and all that.

    buffer sizes you play w/. low for tracking (few efx) high for mixing (efx as u will).

    at the studio i'm currently using direct monitoring, a mixer w/ dsp (in-audible (to my ears) latency efx) digital mixer, at 44.1, and a very high buffer size. it's a bit outdated, but works very well, as my cpu doesn't know when i thro some efx on in the board.

    your current cpu is more than adequate to monitor w/ effexs, and way more than adequate w/ direct monitoring.

    CPU optimization is a must/trial-error do process. but the information is right here for free on this site. check the sticky on the recording computers thread. you basically tell/commit to your computer doing nothing but turning on, and running your audio stuff. if you want to internet surf and stuff use your old one. or make some compromises in your current tools tweaking. there are sooo many variables so, it's always trial and error.

    there's no saying that you got really ripped off, bios tweaks are involved in pc tweaks, but audio specific has a certain set of requirements, not least how your fans respond to yours computers temperature. like music and most people, cpu's like it cool.

    this should be a decent 101 for ya, there's people around this forum who are amazing at daws/computers, i suggest you cover your basics, and read some of the computer based threads. they know way more about the technical side of computing.

    best wishes, keep your patience, now you only have to pay for tweaks w/ your time, and you gain a better working cpu, and you even better, get some knowledge/experience in AUDIO based computers.

  3. haus

    haus Active Member


    Wow man I don't know what to say. This is the best response anyone has ever written in any forum regarding this issue. Thank you so much for your time and effort here - I'll start going through what you've typed in depth.

    For starters all audio effects and processing will be taking place in a preamp - so it looks like I would want to go for direct monitoring? Is there a way I can make this happen for starters using just the motherboard line-in with the setup I have now? I guess what I'm asking is how I go about "picking" direct monitoring?

  4. haus

    haus Active Member

    Or are you saying I need a new external (I'd prefer external at this point) sound card that specifically allows direct monitoring?
  5. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    yeah. you need a peice of hardware/software that allows that. for instance, in my home i use an m-audio fw1814, and reaper. it's just fine, and i used to use it to jam to dropbox songs i had to practice w/ the band i was in. at the studio its a couple motu 2408's. the interface needs to have the ability (option) to basically bypass it's converters (or your ability to hear them). again hevy-er people around here may help more, i'm just trying to keep to basics. so yeah you need to have a peice of hardware that will allow you to hear the analog, while printing the digital and not having you hear it.

    so direct monitoring, is what ya need in an interface, for that. it basically does 'realtime playback' of the tracks, and realtime playback of your input.
  6. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    it could be possible on a stock card but it depends on the software, i used to do it back then in n-track, w/ a soundblaster, but is't been a while i'd have to like figure it out again. i live for this stuff, glad i can aid progress. best wishes
  7. haus

    haus Active Member

    So it looks like at this point my first step would be to grab an external soundcard for this purpose, is that what you're saying?

    Based in my system specs above, what is a quality card that you would recommend? I plan to further evolve this rig into a dedicated recording PC, so I'd like to spend money on quality items as we go... not CRAZY amounts, but anyway - what I'm saying is, what is a solid, reasonably priced interface or card in your opinion, and is that where I should start?
  8. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    yes. it's where you have to start when recording. audio interface is the term. its basically a soundcard. but audio interface is what its called. in the basic type interface that is capabale of a full band recording live i'd go w/ this Product: US-2000 | TASCAM this is in my opinion, the best entry level interface i've tried. it's better than used digidesign stuff. in the typical 18 input 4 out class. better than my m-audio thing.

    presonus makes something like that, they make some nice sounding stuff.

    My personal feeling is that MOTU MOTU.com - Products makes the best bang for your buck interfaces. they are competitive w/ the most expensive in sound quality, and the entry level pricing. if you have a grand get the 896mk 3, it's got dsp. any interface by them will be good. i use the 2408mk 2's downs at the studio, which is a pci type connection.

    if you want to commit to some high end stuff the RME fireface 800 has been tried and true for alot of people, i've never used it. same w/ the Orpheus by prism sound.

    State of the art interfaces are the universal audio apollo, and apogee ensemble / symphony, although the ensemble is kinda old. the symphony and the Orpheus interfaces have options for thunderbolt, but thunderbolt is just a baby right now, and my personal feeling is that there are gonna be new circuit designs based on thunderbolt's capabilities, so i don't think its a big deal.

    please, let us know what the budget is, as well as what type of recordings you'r trying to make (full band live, hiphop?), as well as whether you need mics stands cables as well, or you have that. that would really help narrow down the general suggestions i made.

    you don't want a soundcard, you need an audio interface. there are soooo many, some are mixers as well, so take some time before your buy. lets figure out what you need, and expect to need, as well as what you want.
  9. haus

    haus Active Member

    GREAT response again! NOW I feel like I'm finally getting somewhere. I've been looking over the links you sent me and other options as well. It seems like the USB / Firewire debate is pretty split online. If you had to go with one over the other, which would it be?

    Essentially what I'm doing is simple scores for videos for myself and clients. I don't need anything super high quality, but I'd like to be able to put together stuff that came across really well. I think I prefer the mixing take place after the recording of the tracks - and I don't think I'll need more than 2 inputs at a time ever for this setup.

    Budget: $400 - but the most economical interface that works well will get my vote.

    Software: Acid Studio 8

    *All effacts and tones would be generated by physical gear. I am not opposed to modeling software, but this is something else I know nothing about...

    Your thoughts?
    Thanks again for everything so far!
  10. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    the motu traveler has both connections. i have no problem w/ firewire's performance on hard drives, or interfaces. but it seems to me it's going to be fazed out by usb, and maybe thunderbolt,.? in that case i'd probably go usb, just because those ports are more likely to be around than FW, i guess. any of the ones i put a link to would do fine. i kinda don;t have an opinion on that lol, i think they both work fine, but firewire is most likely to not be future compatible.

    both formats are capable of 18in multi- out configurations, all you may need is the RME babyface, their answer to the apogee duet. on the cheap, fullest i/o, the tascam is gonna be tough to beat for a laptop. the MOTU traveler seems like it'd be cool. i'd probably get the traveler or tascam, if i was in your situation.
  11. haus

    haus Active Member

    Ok, so I'll probably order one of the other units tonight - and what does the RME Babyface do? Please excuse my ignorance, and thanks again for being so helpful with all this!!!
  12. DonnyThompson

    DonnyThompson Distinguished Member

    Don't be mistaken into thinking that an affordable audio I/O can't give you absolutely great results... because it can. Yes.. absolutely... RME and Prism make great stuff. No doubt about it.

    But... don't think you absolutely have to have something in that price class to get great sounding audio.

    Having an understanding of gain structure and other fundamental audio recording principles like mic technique, for example, will absolutely make a huge difference in the quality of your tracks.

    Kmetal was absolutely correct in that there are sooooo many audio I/O manufacturers and models available, and he also brought up an important point regarding the type of connectivity format of the device ...
    Yea, firewire is still around, but I'm not sure that I would bank on it being an available format on I/O's for much longer. USB seems to be the most widely available and accepted connection format these days.

    And I'm sorry to hear that you got burned by a PC "magic man". Yeah, these guys can fix technical stuff like registry issues and virus elimination, but don't ever trust one again to tweak your computer for
    audio production, (unless they happen to be a computer guy who is also a recording musician themselves)....

    Most of your average repair guys don't know anything about what it is that we do. To them, audio is for gaming and playing back MP3's. They haven't a clue about latency, quality of converters, buffer settings for playback and recording... it's not their fault, per se', they just don't know. Although, I'd be pretty upset if a repair guy claimed to know about the many different parameters involved in audio production and actually didn't while charging me a bench fee to optimize my PC for online gaming... LOL.


  13. haus

    haus Active Member

    Thanks Kmetal and DonnyThompson - this forum is extremely helpful. I cannot express to you how helpful this has been - and I have no idea why my old PC seemed to record audio like it did, but it explains why I have not been able to wrap my brain around add-on recording interfaces.

    I went ahead and just snagged one of these packages. I'm sure it's not the greatest but it should get me up and running so I can upgrade as I go: PreSonus Audiobox USB Recording Package | Musician's Friend

    So the main outs on the unit are for the monitors, which are one of the ways I can keep track of what I'm playing in real time. The USB out is what sends that signal to the computer, and then the recording software - once adjusted - accounts for latency and places the track where it needs to go. So no sound as far as what I'm tracking at any given time will be coming out of the PC - is that correct?

    Also, in both of your opinions, what would be the next best thing for me to look into purchasing for better sounding recordings? What should I research next?

  14. haus

    haus Active Member

    Thanks Kmetal and DonnyThompson - this forum is extremely helpful. I cannot express to you how helpful this has been - and I have no idea why my old PC seemed to record audio like it did, but it explains why I have not been able to wrap my brain around add-on recording interfaces.

    I went ahead and just snagged one of these packages. I'm sure it's not the greatest but it should get me up and running so I can upgrade as I go: PreSonus Audiobox USB Recording Package | Musician's Friend

    So the main outs on the unit are for the monitors, which are one of the ways I can keep track of what I'm playing in real time. The USB out is what sends that signal to the computer, and then the recording software - once adjusted - accounts for latency and places the track where it needs to go. So no sound as far as what I'm tracking at any given time will be coming out of the PC - is that correct?

    Also, in both of your opinions, what would be the next best thing for me to look into purchasing for better sounding recordings? What should I research next?

  15. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    nice, presonus makes surprisingly good stuff for their price range. i'm not sure if that interface has direct monitoring, if it doesn't, your audio will have to make the round trip. so this is where the buffer size comes in. use as low as a buffer setting when tracking, is this will keep latency to a minimum. when you start mixing set the buffer size as high as possible, this allow you the most processing power.

    the next place to start looking for recording gear is acoustic treatments. i made my own for the studios, and i just bought some aurelex foam for my listing room at home. the importance of acoustic treatments cannot be stressed enough. the best most high tech speakers don't matter if the room doesn't reflect bass and treble relativity evenly.

    so what happens is you end up making this mix that sounds beautiful in your studio room, and you take it out to the car and it's exaggerated in some way. maybe the bass is way higher or lower, suddenly it sounds like there's too much reverb, or not enough. so gonna need some absorption on the front wall to the left and right, the ceiling area above your head, and the rear wall. and as much bass trapping as you can afford/have room for. that's a good place to start. glad i could help.
  16. DonnyThompson

    DonnyThompson Distinguished Member

    From what I'm able to see when I zoomed in on pics of the front and back of the Audiobox, I believe you should be able to direct monitor, and here's why...

    On the back, you'll notice your TRS outs, but you also have a headphone jack. On the front of the unit, you'll notice an adjustable "mix" feature... this allows you to monitor what is coming from your PC, or monitor what is going to your PC...or a mix of both.

    The "input" side of the control will let you hear what you are inputting into the audio device in real time. The "playback" side of this control will allow you to hear either what has already been recorded, or will allow you to monitor your soft synths as you play.... BUT.... you still may encounter latency when monitoring the playback.

    Presonus claims "zero latency", but this will be determined by several factors...

    The first is the CPU/Ram power you have on your computer.

    The second is how that input/playback "mixer" function is set.

    If you are recording a track using an internal soft synth, there is no "input" per se', because you are triggering those samples which reside inside your PC/Recording Program... therefore, you will need to monitor the "playback" to hear them. These internal sounds aren't like vocals, or another acoustic instrument which you play in "real time", so you'll need to rely on the "playback" side of the function to hear those samples as they are being recorded.

    You may notice a slight delay from when you strike a key on your controller versus when you hear that sound played, is what I'm trying to say.

    Some latency settings can also likely be adjusted internally within the Presonus mixer function, which would let you adjust your RAM allocation to that device. The lower the RAM buffer, the less latency you should encounter, but the higher the possibility for glitches: freezeups, dropouts, stutters, knocks, pings, moos and quacks.... LOL...

    Conversely, the higher the buffer setting, the more latency you will encounter, but with a lower degree of possible error.

    Kmetal mentioned in another post somewhere here on the forum that by and large, it's a good idea to use lower buffer allocations for tracking, and higher values for mixing, and he gave valuable advice on this.

    BUT....What you need to do...and what we've all had to do, is find that "magic" setting that allows for the most optimum performance of both recording and playback... finding that balance that allows you the best of both worlds.

    Don't be surprised if you have to tweak things a bit, is what I'm saying.

    Let us know how it works out for you.

    Good luck :)

  17. kmetal

    kmetal Kyle P. Gushue Well-Known Member

    yeah +1 on the magic settings. trial and error is the only way.
  18. haus

    haus Active Member

    Hey guys,
    Well I wanted to come back to this thread so I could let you know that the gear WORKS. I have been able to lay down some scratch tracks (in mono, not stereo - which I believe is based on the I/O unit alone), and for starters that is all that I needed. My next question is this: what's next? I read room treatments above, and I will get into that as I start to record live amps and vocals, and get more serious about the mixes... but after that, what would you recommend I look into next?

    A few other basic and super important questions:

    1. What is the point of mixing, and what are some tips for the best mix possible? (Feel free to link to other threads)

    2. What is the point of mastering? Can you do mastering at home on some level? (Feel free to link to other threads)

    3. The studio monitors that came with this package sound flat - which is the point, right? What is their purpose, is it just to normalize sound as much as possible? When I switch over to the main speaks as the playback device I gets TONS of bass... probably because they have a bass knob that exaggerates things, yes?

    Thanks a million!!

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