Your Vinyl Transfer Workflow (sharing best needledrop practices)*

Discussion in 'Audio Hardware' started by Vocalpoint, May 11, 2011.

  1. ghost rider

    ghost rider Forum Resident

    Location:
    Bentonville AR
    Thanks for your input maybe I can use a friends disc to test out w8.

    I loaded up W7 off one of my disc and it didn’t work much better. I also sent a message to Izotope and they suggested that it was either the CPU or RAM. I did some research and found 4 more gigs of the same ram I built the computer with on eBay used but I can return it if it won’t work. The fact that it’s the exact memory, so it will be 4 2gig sticks all the same suggest that it will. Maybe 8 gigs will do the trick.
     
  2. Stefan

    Stefan Senior Member

    Location:
    Montreal, Canada
    Grant, I thought I'd update you again on my progress with this issue and once more post it in this thread in case someone else runs into the problem.

    I found a work-around that seems to resolve the problem nicely. My motherboard has an "optical S/PDIF" out that supports up to 192kHz. My E-MU card has both coax S/PDIF ins and optical (the optical in and outs are labelled ADAT but they work as well for optical S/PDIF. So I set the default Windows playback device to my Realtek digital output. Then in E-Mu Patchmix, I added a strip for the Card S/PDIF input. You have to be sure to set the inputs in Patchmix for S/PDIF to coax disabled, and most importantly, you cannot have the optical and RCA/coax inputs or outputs connected at the same time. Once I did this, I was able to get Windows and internet sound running perfectly through the E-MU card for recording and playback.
     
  3. Grant

    Grant Life is a rock, but the radio rolled me!

    Thanks for this! I'll try it.

    New problem: thinking I had my audio editor problem sorted, I sat down this morning to do a needledrop of some records I bought last night. But, I can't get any programs to record in stereo. When I look at the routing in Audition, Sound Forge, and Reaper, the left E-mu card input says "(mono)", and I can't get rid of it. So, I get nice recordings, but it's all in mono.:realmad:
     
  4. Paul H

    Paul H The fool on the hill

    Location:
    Nottingham, UK
    I opened Audition for the first time in my life last week when it was suggested to me that I should use it to make the recordings. I'm not at all familiar with it.

    I used Automatic Click Remover from the Restoration tab. I adjusted the settings to try various things but nothing worked.

    I tried something similar in Audacity (I tried matching the peak amplitude until I realised that the pop is much louder than the rest of the track so it distorts the value). I'll have a go using this method, but would the pop affect the RMS?

    The Asus has a volume control on the card itself that can be manually adjusted (it seems to adjust in roughly dB intervals) and there is a software volume control. Neither had changed. Thinking about it, there's also a volume setting in Windows but why would any of these settings change without me adjusting them?
     
  5. Grant

    Grant Life is a rock, but the radio rolled me!

    OK, I see. Yeah, Audition is a very comprehensive program to master. I learned it by trial and error, with the help of their help section. I even have two owners manuals. But, through using the program, and gaining experience, you find all kinds of unorthodox ways to get things done. So, in that in mind, I don't have much luck with the Click/Pop Eliminator. I have found that you can literally cut out (delete) a back click or pop by zooming way in, selecting only that spike, highlighting it, and then hitting the delete button on your keyboard. When you zoom back out and play that section, it is most likely that you will never hear the pop, or anything else missing. That is actually my last resort. I try to use the excellent methods in iZotope's RX first.

    Syntrillium originally owned and developed Audition (which used to be called "Cool Edit 96", "Cool Edit 2000", and "Cool Edit Pro". Their philosophy was for the user to figure it all out. No joke.
     
  6. Paul H

    Paul H The fool on the hill

    Location:
    Nottingham, UK
    I had tried removing as much of the pop as I could with Audacity and then cut and paste a couple of samples to cover the damaged section but it just didn't sound smooth at all. There are other settings for eliminating clicks and pops but at first glance little of it made sense. When I get a chance I'll look into the capabilities of Audition a bit more but, of course, it's always more tempting to stick with what you know (Audacity).

    Re: Cool Edit Pro: well, I never knew that...
     
  7. Paul H

    Paul H The fool on the hill

    Location:
    Nottingham, UK
    A thought just occurred to me (which I can't test right now) that the Asus software may be assuming that the joint Mic/Line in connection is used as a Mic and has that set as it's default. If, therefore, I've disconnected the Asus (either at the laptop's USB port or at the Mic/Line in connect on the Asus itself, it may be defaulting back to Mic.

    My guess is that that was what caused the massive input volume overload I experienced. When I get a mo (which may not be for a while - I have the small matter of my impending marriage to attend to!) I'll check this theory out.

    Out of interest, assuming that the settings are adjusted correctly for each, does it matter whether the Asus thinks it's getting Mic or Line in input when I do a 'drop?
     
  8. ascot

    ascot Senior Member

    Location:
    Wisconsin
    It might affect the highest peak but that depends on how much of a spike you have compared to actual music.
     
  9. Grant

    Grant Life is a rock, but the radio rolled me!


    If you can send me the file, I will see if I can fix it.
     
  10. Stefan

    Stefan Senior Member

    Location:
    Montreal, Canada
    I used to have this problem with Audition 3.0 in that by default, it lists the left mono channel. If you check the dropdown list in the Audio Hardware settings, there should be a listing at the bottom with an S in it. that will give you the stereo input. It's been years since I used Sound Forge so I can't help you there, but I use Reaper all the time and I just upgraded to version 5. It also defaults to a mono input for a track unless you set up a layout then save it as a project template and then in the Preferences, set that project template to be used whenever creating a new project.

    To set the track in Reaper to record in stereo, first make sure you've armed it for recording (click the red button), then in the mixer, right click on the volume sliders, and towards the bottom of the resulting context menu, you should see an Input entry with an arrow expanding a menu to the right, where you can choose stereo inputs.
     
    Grant likes this.
  11. Stefan

    Stefan Senior Member

    Location:
    Montreal, Canada
    that may be what's doing it. However, I leave my U7 connected all the time and have never had the volume jump like that. It might have happened since you were trying different settings, etc. Once you get a sample rate and everything else set up, you should be able to leave it set up and not have any problems with settings changing.
     
  12. Paul H

    Paul H The fool on the hill

    Location:
    Nottingham, UK
    That's what I'd hoped :)

    Perhaps I disconnected the Asus from the laptop when I put it away. Any thoughts on the differences between Mic and Line In when using the same connection? Assuming I can get the default settings "right" for each input type, would it actually make any difference or am I better making sure I'm using the Line In option?
     
  13. Stefan

    Stefan Senior Member

    Location:
    Montreal, Canada
    I would definitely use Line In as it's designed for the signal level you'll be getting from your phono pre-amp.
     
  14. back2vinyl

    back2vinyl Forum Resident

    Location:
    London, UK
    I'm not familiar with the Asus at all so I've no idea what I'm talking about. But just Googling the manual, there should definitely be a user interface for the device and that's what I'm suggesting you might try checking. After installing the driver and rebooting, you should apparently find an icon for the Xonar U7 Audio Center in the system tray at the bottom right-hand corner of the laptop screen. If you double-click on this, it should open up the Audio Center where you'll find pages of menus to go through and if it was me, I'd be going through these very carefully indeed to make sure everything was set up right.

    You probably know all this already so apologies if I'm just telling you stuff you already know.

    About the settings in Windows - again, I'm very far from being an expert but there is this thing called shared mode that you need to be careful of. It's quite common for Windows to be reproducing sound from two different sources at the same time and when it's doing that, it's operating in shared mode. It's also quite common for these different sources to have different sample rates and bit depths and that's why, on the advanced tab in the sound control panel, you're invited to decide what sample rate and bit depth Windows should insist upon when operating in shared mode and Windows will then order your device to resample everything to meet your chosen parameters. (Maybe it balances volume, too?) You can see that this does give Windows a degree of control over the recording process and in order to take back that control, you need to ensure that you're operating in exclusive mode, not shared mode, when recording. So you need to go to the advanced tab and check both the exclusive mode boxes and you probably need to ensure also that you've given Audition exclusive access to the Asus and that Audition is set to record in the same resolution as you've told the Asus to deliver, or you'll get into a real mess!

    You'd probably find this a lot easier with more up-to-date software than Audition 3. You could try a free trial of the current version of Audition to see where you get with that. There's also Vinyl Studio, which is custom-made for needledroppers and costs $29.95 - I think you can get a free trial of that, too. I think it's excellent.
     
  15. Paul H

    Paul H The fool on the hill

    Location:
    Nottingham, UK
    OK, so I think I'm up and running. I have an album 'dropped as a single file. I'm familiar with declicking (at least in as much as I do it manually working on the clicks I can hear) but I'd appreciate some advice about other post-production work I've seen mention, specifically the following:

    1) High-pass / low-pass filtering. I understand the gist that they filter out frequencies but I'd like to know (in layman's terms) what the purpose of this is and how I should go about doing it (using either Audacity or Adobe Audition).

    2) Converting from 32/96 to aac lossy for use on my iPod. I'm happy to keep the files at 32/96 for playback on my main system (assuming that Foobar and/or Winamp are happy with that level of resolution). I've seen mention of needing to dither and resample in a particular order but previously I've just ticked the dither box, selected the appropriate aac format (320kbps) and let Foobar do the work.
     
  16. One_L

    One_L Forum Resident

    Location:
    Lower Left Coast
    For converting the files to play on an iPad (mp3?, AIF?) I use Adobe Audition. Add your files into Batch Processing, there you can adjust the Export Settings (little box a that bottom of the window) I use Audition to dither and resample although I'm guessing others have a more specific protocol to follow. I'd be happy to hear other choices for dither and resample as well.

    Can't help you with the High/Low Pass settings.
     
  17. Bathory

    Bathory 30 yr Single Malt, not just for breakfast anymore

    Location:
    usa
    Ortofon Arkiv,all my troubles were solved. Recording/burning at 192khz. Make simple burns to Taiyo yuden cd-r's
    I also use ortofon nightclub mkII. GREAT HIGH OUTPUT AND FREQ RESPONSE.
    I leave many of the slight crackles/pops intact, adds a lot of goosebumps hearing them, and it's at a mega good digital recording, adds a little something to the music. Don't say pops& noises either smarty asses)
     
  18. superstar19

    superstar19 Authentic By Nature

    Location:
    Canton, MI, USA
  19. DickLaurentIsDead

    DickLaurentIsDead Forum Resident

    What's the best way to go about removing that rotational hiss noise you get when there is a scuff?
    They aren't recognized as clicks as they don't really cause spikes. And they are really hard to identify even when zooming in on the file in Audacity.
    But the noise is definitely there.
    The decrackle feature doesn't seem to help in Clickrepair, unless I'm doing something wrong.

    Any ideas?

    Looks like the Denoise software might help at Clickrepair?
     
    Last edited: Sep 7, 2015
  20. vinyldoneright

    vinyldoneright pbthal

    Location:
    Ca
    Post a sample of the noise
     
  21. DickLaurentIsDead

    DickLaurentIsDead Forum Resident

  22. vinyldoneright

    vinyldoneright pbthal

    Location:
    Ca
    Those you need to manually fix them one by one
     
  23. DickLaurentIsDead

    DickLaurentIsDead Forum Resident

    Really?
    I guess they're stayin!
    Thanks.
     
  24. Paul H

    Paul H The fool on the hill

    Location:
    Nottingham, UK
    Bump
     
  25. Stefan

    Stefan Senior Member

    Location:
    Montreal, Canada
    Some folks do, others don't. Any filtering will affect the audio, the question is whether it will be audible to you. We can get all "purist" and insist on the pure, unsullied majesty of the raw file, but unless you're working in a studio as a professional on material that's to be sold to others, I wouldn't worry too much about it. This is a hobby and it should be about producing sound quality you enjoy.

    Personally I wouldn't worry about low-pass filtering unless you had super high-amplitude high frequencies that could damage tweeters, which you simply won't get with vinyl playback. And, chances are unless you're a bat or a dog, you can't hear whatever is up there anyway!

    I would (and do) apply high-pass filtering to remove subsonic rumble and resonant frequencies. If you spend enough time with a quality spectral-based audio editor such as iZotope RX, you'll discover there really isn't much down there. There are exceptions, but I rarely see any audio material below 30Hz, and on many LPs, there's nothing really down below 50Hz or higher. Whether this was rolled off during the recording or vinyl cutting process, or my system simply isn't picking it up, if there's no musical information there, there's no point keeping all the crud that is down there as it will just make your woofers or subwoofers work for nothing, which will muddy up the bass that is there.

    Your approach is the best one. In theory, whenever one is reducing the word length of a PCM audio stream, you should apply dither to reduce/avoid distortion. In practice, it's not really all that essential for reducing from 32 to 24-bit audio as the dither and distortion are both well below the threshold of hearing. It is more important when reducing to 16-bit audio as the distortion from not dithering can be audible. It's not very audible and in fact some folks can't really hear it at all, but it is above the minimum threshold that's been defined by science as being perceptible to humans.

    Plus look it this way, given all the other noise you're getting when needledropping (vinyl background, etc.,), the tiny bit of hiss you add using dither is quite trivial. Don't worry about noise shaping, etc., in fact you're better with flat dither if you plan on processing or converting the files any further.

    I'd also keep your final source files in 24/96 not 32/96 as that's a waste of space. There isn't a lot of equipment available to home users with 32-bit file playback that isn't really in the background changed to 24-bit word length. In fact, the better home playback digital to analog converters only offer the equivalent of about 19 to 20-bit audio (-114 to -120dB). Plus if your files are 24/96, it makes it easier if at some point you want to create audio-only DVD's, homemade DVD-As, etc. Plus some equipment doesn't accept 32-bit files. You might be ok with PC playback for now, but perhaps at some point you'll pick up a receiver that streams audio through your network.
     

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