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Thread: Anybody building a new PC DAW System

  1. #31
    Mentor dave.king1's Avatar
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    In Reaper you are listening to quite a naked sound and a single sound file for a drum kit getting it sounding right is quite a challenge because each drum has different EQ & Compression requirements to sound good.

    I use Drummica from Senheisser which is free, a bit of stuffing around to get running but it is recordings of a live drummer with a huge library of beats.

    Drummica here http://de-de.sennheiser.com/drummica the site is in German so a bit of thought is required when working through the download but it is fairly intuitive

    I sit down with my band mates and find the pattern and sound we want and then come home and create the drum tracks.

    Typically kick ( front & back ), snare ( top & bottom ), hi hat, overhead & toms if applicable.

    These are all individual tracks that need to be aligned, kick on 1&3 snare on 2&4 etc which takes only a couple of minutes , I then apply EQ & Comp to each track to get the sound I want, slight verb on snare, I balance them individually and then route all these through a drum master track so I can change the entire kit volume in the mix without trying to rebalance each item each time.

    I also use automation to bring the snare and cymbals up where required.

    Apart from the drum VSTi I don't use anything to treat the drums that isn't native in Reaper.

    Kenny Gioa has a heap of terrific Reaper specific tutorial vids on YouTube.

    Your main production tools are EQ and Compression and remember "less is more" with these tools, chaining a couple of EQs using small steps will give a better result than smashing it down unless you are looking for a specific sound

    Keep asking and I'll do what I can to help.
    Last edited by dave.king1; 15-08-2017 at 07:59 AM.

  2. Liked by: Guvna19

  3. #32
    Mentor dave.king1's Avatar
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    If you are really keen you can also program your drum parts from scratch in MIDI and then load a VSTi such as GTG DPC3.

    http://www.vst4free.com/free_vst.php?id=385

    I've done it, bloody long process but it does enable you to get what you want if you have odd time signatures
    Regards
    Dave King

  4. #33
    Member Guvna19's Avatar
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    So far managed a basic drum beat today in reaper using MTpowerdrum, great fun to jam along with.

    when i add guitar for recording, i get a little snap crackle pop? not sure if i can change setting to correct?

    PC is good, interface is good, average monitors, - maybe its the monitors i dunno.

    dont like using my amps for drum parts.

    any advice? Guv

  5. #34
    Overlord of Music Simon Barden's Avatar
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    Normally it's because the audio processing stops for a short while as the processor is too busy doing other things. Increasing the buffer size on your interface will normally stop this - but at the cost of increased latency.

    How much you can do on a DAW without having to increase the buffer size mainly depends on processor power and the drivers for your audio interface. More RAM can help, as can having an SSD as the system drive, but most of it is down to processing power and well written drivers.

    However, there are other things that can also take up processor time - things like 3rd party virus checkers, wireless networks and windows tasks running in the background. There are also programs that when sequenced to run by Windows simply take a long time before they pass back control to Windows, during which time your audio buffer may have filled up without being processed, resulting in an audible glitch. There are various latency checkers you can download, I've used this one http://www.resplendence.com/latencymon which gets a good recommendation from a fairly large UK PC building company that I know the owner of. If it highlights a program with a high latency, then you either need to shut it down (if non-essential) or find an alternative.

    This is one reason why DAWs are still best used as dedicated machines (or at least dual boot).

    Also, if you don't have a separate video card but rely on the on-board display, then that can take up a fair bit of processing if the DAW windows are scrolling.

    Also check any power saving options - especially if it's a laptop. You want everything running at full power all the time with a DAW. you don't want windows deciding to ramp your processing power or hard drive speed down.

    So if the PC is multi-purpose, you may need to set up a dual-boot system with minimal programs loading for DAW use - or at least a separate profile. There are some programs, like Adobe's Acrobat Reader, which constantly contact the web (at least it did a few years ago when I last used it). Also beware of any updater programs that keep looking to see if there is a new version available.
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  6. #35
    Mentor dave.king1's Avatar
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    Thanks for chiming in Simon I've been hoping you would.

    One of the problems I've had with VSTi in Reaper is random peaks the one I use for Hammond B3 does it with peaks way over +20db and this auto mutes the track and another piano VSTi does what you describe and it's often not noticed until the project is rendered.

    The solution is to go to the JS plugins and apply the Master Limiter to the track in question, set it to about -6db this will control transients on the virtual input but won't stop you from getting the output where you want it.

    I don't know if this issue is unique to Reaper or more likely the VSTi that I am using.
    Regards
    Dave King

  7. #36
    Overlord of Music Simon Barden's Avatar
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    Probably the VSTi. With the floating point maths used in DAWs, they've got a huge theoretical dynamic range (around 1680dB), so peaks of +20dB aren't a problem at all. But that's not to say that a VST/VSTi plug-in or Reaper itself doesn't have some default protection to save speakers etc. in the event of detecting a particularly loud transient. I'm a Cubase person so not really familiar with Reaper, though I have tried it a couple of times.

    Many VSTs modelled on real FX units behave in the same way to incoming signal levels, so levels above the system nominal 0dB will often result in distortion or severe limiting/compression.

    Obviously the trick here is to carefully set the gain levels of instruments so that they don't go above 0dB, and generally have a much lower average signal level than this.
    Build #1 ES-3 under construction http://www.buildyourownguitar.com.au...ead.php?t=6489
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  8. #37
    Member Marcel's Avatar
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    I'm just trying to contemplate what 1680dB equates to in real life......

    It's like going from way beyond the quiet silence of outer space to less than 1cm away from ground zero of a 100GT nuclear blast !!!

    That IS a huge (mathematical) range...
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  9. #38
    Overlord of Music DrNomis_44's Avatar
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    Another good latency tester app is one called DPC Latency Checker, it's free and I sometimes use it to fix DAW latency problems after I've tried various buffer settings, here's a link to the website where you can download it from:

    http://www.thesycon.de/eng/latency_check.shtml

  10. #39
    Quote Originally Posted by Marcel View Post
    I'm just trying to contemplate what 1680dB equates to in real life......

    It's like going from way beyond the quiet silence of outer space to less than 1cm away from ground zero of a 100GT nuclear blast !!!

    That IS a huge (mathematical) range...
    Something to remember is to keep input levels below 0dB when tracking (peaking at -12 to -18 is considered fairly safe), and output levels below 0dB when exporting to a finished .wav or whatever. Once "inside" the DAW, while the audio is being handled by floating point math, it is safe to exceed 0dB.
    Last edited by Paul_H; 18-08-2017 at 08:44 AM. Reason: typo

  11. #40
    Overlord of Music Simon Barden's Avatar
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    You really need a 'true peak' meter to set the master output level when exporting to .wav files and the like, and even then set the maximum output using the true peak meter to -0.5dB or so. This is because inter-sample peaks can push the real output up by up to 3dB.

    You don't need to worry whilst it's still it's in the DAW and Paul_H said (though as I've previously said, some plug-ins may give you a lot of distortion or saturation if the incoming signal is too hot). So it's still best to keep everything below an indicated 0dBFS and keeping individual bus output levels to between -12 and -18dBFS. If you're working at 24 bit depth, then at -18dB you've still got far more theoretical dynamic range available 21 bit's worth) than a 16-bit .wav file has, so you're not giving anything away by working with a -18dBFS signal.

    Inter-sample peaks are often the reasons that MP3s sound bad, especially if they've been taken from a .wav file that has sample peaks at 0dbFS or just below.
    Build #1 ES-3 under construction http://www.buildyourownguitar.com.au...ead.php?t=6489
    Build #2 EX-1 (with alder body) Yet to start. http://www.buildyourownguitar.com.au...093#post123093
    Build #3 Customised ES-1 for an ES-330 style build. On order, expected November.

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