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I'm having a little bit of trouble with my setup. I'm using a Mackie Onyx Satellite and M-Audio AV-40 monitors connected with TRS leads. Not the most Stellar setup, I know. I'm mixing some vocals I did with a backing track and I start to notice the odd bit of distortion in the sound. It's very intermittent, and seems to be happening more at certain frequences than certain volumes, but maybe it's a combination of the two, on certain notes of a piano part of example. I've just spent ages trying to troubleshoot it and I've finally noticed that if I turn down the Control Room Out volume in the interface, it goes away, even if I then turn up the volume on my monitors up so that everything's then at the same volume it was before, or even louder. This is somewhat of a relief as I was starting to worry that my monitors were blown, but I am somewhat confused. The manual that came with my monitors said that when connecting a line level source to turn it to max, and I was reasonably sure the Control Room out was line level. I only have to back it off to about 75% before the distortion all but goes away. What could be the problem here? Is the inteface putting out a slightly hotter signal that it should, or are the monitors not able to accept quite as hot a signal as they should? It kind of seems like one or the other must be the case. edit: I tell a lie, it's more like 50% to get it to go away completely. edit#2: I told another lie. It's actually the interface's manual that told me to have the Control Room out as close to Max as possible. quote:Use this control as close to the MAX (fully clockwise) position as possible to experience the best audio quality. Turn down the input sensitivity control on your active monitors or your passive monitors’ power amplifier in order to turn up the CONTROL ROOM LEVEL knob as far as possible. There is no input sensitivity control on the monitors. So I guess I should just have it as loud as I can without distortion? Will I have damanged my monitors having it on Max all this time? It's weird that I can rinse certain tracks with heavy bass and things and not notice any distortion, but then a solo piano part can sound distorted. 
chippy fucked around with this message at 20:42 on Mar 30, 2011 |
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Mar 30, 2011 20:05
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Looking at the manual, it appears that Control Room out actually shares its signal with Phones 1, meaning you are plugging your monitors into a hybrid headphone out. A headphone amp can certainly clip a line input. Doesn't sound like you did any serious damage if you can crank them loud without having the problem.
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Mar 31, 2011 00:18
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I thought it only shared it with headphones 1 if the pod wasn't docked in the base unit. If you use the pod un-docked they share one volume control but if it's in the base unit they have one each, the speakers are plugged into dedicated L/R control room outs on the base and the headphones into a headphone jack on the top. Or did you mean something more technical than that? Anyways, yeah the speakers do seem fine. I still find it a bit strange that I could play a loud, commerical track and hear no distortion but one channel playing in Cubase peaking at -8 or something and I'd heard distortion. Although do be fair some commercial music was doing it too. Blasting Squarepusher sounded awesome and no noise but something like Seeed (dancehall/reggae type stuff) sounded loving awful. That was when I first noticed it, I really thought I'd blown them then. edit: Just popped the pod out for a look. Even undocked they have seperate physical outputs but they share a volume control, so I guess that means sharing a signal, but I'm not sure that's the case when it's docked though. chippy fucked around with this message at 01:05 on Mar 31, 2011 |
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Mar 31, 2011 01:02
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The block diagram makes it look like the separate outputs on the Satellite itself are fed by the same gain stage before they split to the output jacks. You're right, I forgot about the mirrored outputs on the base station, they do seem to have a separate gain stage from the CR outputs on the Satellite itself. It still isn't clear whether the output stages are the same kind though. Logic would say they made them different, manufacturing might be another story. Is the knob detented at 12 o'clock? It might not mean anything, but on Mackie mixers 12 o'clock is unity and farther to the right is adding gain on every knob but the preamp gain. Also, commercial releases are much more compressed than a raw track in your studio. With any system you are way more likely to blow something yelling into a microphone than blasting a CD, dynamic content probably has something to do with why you noticed it first with raw tracks. ChristsDickWorship fucked around with this message at 01:48 on Mar 31, 2011 |
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Mar 31, 2011 01:45
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There's no detent or anything on the scale round the knob it to indicate that 12 o'clock might by unity, but you could be right all the same. 12 o'clock seem to be the position about which the distortion completely disappears.
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Mar 31, 2011 12:57
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Crossposting my question from the headphones thread. I play an 8 string guitar through a POD HD500 and use headphones with the setup. My 595 modded 555s just are not cutting it. The low end doesn't come through as clear and tight as I would like it to. What headphones could I get that not only would work great with extreme low end monitoring, but great for gaming and listening to music? I'm looking at Denon AHD2000s but that price tag is way way way over what I could spend anytime soon.
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Mar 31, 2011 13:17
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The Shure SRH440, 840, and 940 headphones have a lot of great low end detail that is accurate rather than boomy. The higher up you go, the better it gets.
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Mar 31, 2011 16:34
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Hogscraper posted:You can get some awesome drum sounds out of just a single mic or two. The drums on the new Black Keys record were all one mic mono drums. Sounds awesome. Granted, they ran nice mics into nice pres and really pushed said pres into distortion but it's a great sound. Can you explain what you mean by pushing mic pres when recording digitally? I know next to nothing about hardware, is this not just turning the gain up? Cause if I normalize two recordings done at say 25% gain and 80% on my interface, they sound exactly the same to me. Any higher than 80% and most things start to clip digitally. Do I have bad pres or am I just missing something?
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Mar 31, 2011 17:15
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If you have another gain stage (like a mixer) to attenuate the output of your preamp you can drive it well into distortion then trim the overall level so you don't clip your DAC.
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Mar 31, 2011 18:59
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Ok that makes sense, I knew I was missing something hah. Thanks.
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Mar 31, 2011 19:11
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cross-posting: I'm running an Akai MPK49 into an E-Mu 0404 USB, which then goes into the computer for sounds, and headphones out for my stereo. 1. When I play using Reason, the sound doesn't work in Foobar/Firefox/Reaper, so I can't jam along to songs. How can I change this ? 2. Even using the lowest sample rates, I still get latency - even with only a Redrum. Like just playing a kick at the same time as the metronome and there's a delay which fucks everything up. Help ? 3. I bought a MIDI cable to run between the Akai and the EMU so as to reduce latency, but I don't think it's changed anything - and fiddling with Inputs in reason to make it "Midi In" never gives any sound. I just wanna jam and make bad pastiches of Dilla beats
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Apr 3, 2011 00:27
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An0 posted:cross-posting: 1) most audio apps talk to the sound interface in exclusive mode - once you start up Reason, it has your Emu's full attention, so that other apps trying to play sound won't interrupt the stream and cause latency or stutters. If you want to jam along, you'll probably have to stick the songs you want to play with on an ipod or something, and plug that into one of the lines in on the Emu and route that input to go to the main outs that your computer or keyboard is outputting to (I have no idea how to do this on something like an Emu, the user guide should show you more). 2) make sure Reason is set to use the ASIO drivers for your Emu, not directsound or windows audio; the latter two options are not intended for doing music production and have horrible latency. 3) switching between two different MIDI interfaces won't do anything to resolve latency problems; that's entirely an audio interface issue (usually driver or system related, as above). edit: when you say "headphones out for my stereo", you do mean you have the headphones plugged into the Emu 0404, and not your computer's onboard soundcard, right? Militant Lesbian fucked around with this message at 01:51 on Apr 3, 2011 |
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Apr 3, 2011 01:47
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What I think would be cool is a VST that lets you isolate squared-off frequency bands that lets you choose a range and then takes anything out of that range to -∞ I could simulate this with my Equium or Firium plugins but it'd be nice to have something where I could simply input Hz numbers. Maybe this exists and some kind soul might post a linku-nyo~
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Apr 3, 2011 02:24
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Hey so another question here, google has failed me. Does anyone have experience producing death vocals (a.k.a. growling)? I'm currently engineering an album for my band, and we'll be doing the vocals within the next month or so. I have no experience with recording these types of vocals , but I know a little bit. For instance, I know that doubling the vocal tracks and pitch shifting one slightly is something commonly done in death metal. Any advice regarding mic selection, room type, or effects would be greatly appreciated. Btw, here is a link to our facebook page with our demo from 2009 on it. It should give you an idea as to what we're going for. (I also want to mention that I had nothing to do with that recording besides playing bass.) Schlieren posted:What I think would be cool is a VST that lets you isolate squared-off frequency bands that lets you choose a range and then takes anything out of that range to -∞ I don't think there's anything in one neat package that does exactly what you're describing. A "notch EQ" can cut out a frequency, but its typically a very narrow bandwidth. Here's what you could do , but it won't be pretty. Set the track to having no output, then send the track to multiple buses. Each bus could then have a low and high cut filter on it, with each bus having its own range of frequencies cut out. Then, send those buses to another bus so you can control the volume with one fader. Huge Lady Pleaser fucked around with this message at 20:31 on Apr 3, 2011 |
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Apr 3, 2011 20:16
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HotCanadianChick posted:1) most audio apps talk to the sound interface in exclusive mode - once you start up Reason, it has your Emu's full attention, so that other apps trying to play sound won't interrupt the stream and cause latency or stutters. If you want to jam along, you'll probably have to stick the songs you want to play with on an ipod or something, and plug that into one of the lines in on the Emu and route that input to go to the main outs that your computer or keyboard is outputting to (I have no idea how to do this on something like an Emu, the user guide should show you more). 1) What I'm doing now is just using a lovely free synth in Reaper, and putting a track I wanna jam to in there. Works well enough for me for now 2) The driver is the right one, Asio Emu thingy. The latency isn't horrible, but it's still really annoying. It may be the computer, but it's only 2 years old. It's not just Reason, because I'm getting a latency on Reaper, with just a click and an ultra simple synth. I close all my other programs while either Reaper or Reason is running. This is using the lowest bitrate by the way. So it says that there's an expected 2-4ms latency I think. Am I just hosed ? 3) ok 4) yeah, stereo plugged into the audio out on my EMU. Thanks for taking the time to answer 
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Apr 6, 2011 22:40
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A long time ago, I bought an M-Audio Axiom 49. I bought it to use in conjunction with either a rackmount synth or some vsts or something. I bought the mbox bundle with protools LE 8 recently. I've installed the driver for my Axiom to work with windows 7, but protools doesn't detect my midi controller. I'm using the controller through usb, should I try midi cables instead? I don't really know what more information I should give you, and I would appreciate any and all advice to get the two working together. Thanks. *edit* I don't own monitors, so I'm trying to use headphones, but I'm not getting anything through them. It has been connected to monitors and those worked. But my headphones will not. What can I do to get this to work? *double edit* I think I figured out the headphones issue. Tetris Crowley fucked around with this message at 17:10 on Apr 10, 2011 |
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Apr 7, 2011 04:26
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I don't like to bump threads, but I was really hoping someone here would be able to give some advice. BBW FEVER posted:Hey so another question here, google has failed me. If not, I'll have to venture into one of those audio engineering forums with people who might just be geniuses as producing music but are completely retarded when it comes to forming coherent posts. 
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Apr 13, 2011 20:45
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BBW FEVER posted:Does anyone have experience producing death vocals (a.k.a. growling)? I'm currently engineering an album for my band, and we'll be doing the vocals within the next month or so. I have no experience with recording these types of vocals , but I know a little bit. For instance, I know that doubling the vocal tracks and pitch shifting one slightly is something commonly done in death metal.
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Apr 13, 2011 21:41
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no matter what I do, I just cannot seem to find a decent heavy metal tone, that's not too distorted (yeah too distorted for metal) and heavy enough that it's not just pure treble. I'm usind Pod Farm with the Metal FX pack aswell, so someone tell me where I'm going wrong?
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Apr 14, 2011 15:19
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Equalizer settings?
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Apr 14, 2011 15:32
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Benf199105 posted:no matter what I do, I just cannot seem to find a decent heavy metal tone, that's not too distorted (yeah too distorted for metal) and heavy enough that it's not just pure treble. High-passing guitars always seems like a good idea when mixing. It can be surprising how getting some of that lower mud out really makes room for the bass and the kick and actually makes the guitar sound even bigger and thicker.
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Apr 14, 2011 18:11
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Benf199105 posted:no matter what I do, I just cannot seem to find a decent heavy metal tone, that's not too distorted (yeah too distorted for metal) and heavy enough that it's not just pure treble. Oh yeah, if you like very tight low end don't use a Marshall cab model or anything with greenbacks or v30s in it. Use something modern that's made for that sort of thing like a Mesa or Peavey cab model.
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Apr 15, 2011 21:52
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What are my options for mixing youtube videos when they are converted into Mp3's? I am trying to make a crisp version of this video for a friend. http://www.youtube.com/watch?v=a1o82BVKFp8 Goal is to have something similar to this. http://www.youtube.com/watch?v=C5iTwhWYnzE Now I did fix it up a bit with some EQ, normalization, and some Compression. Is there something else I should consider or am I wrong for doing this since I do not have the original audio files to begin with anyway? It may seem like a silly question since I am a musician, not so much a producer. Less Claypool fucked around with this message at 21:17 on Apr 17, 2011 |
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Apr 17, 2011 21:14
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gingivitis the wart posted:Well the Shure SM7b is a goto for a lot of metal producers, it works well for in your face vocals without a lot of dynamics. Make sure your doubles are tight so the vocals don't lose their punch or become incomprehensible. Mixing in a little saturation (like tape distortion, Decapitator is great) can help but don't overdo it. There are a lot of people without much recording experience think that modern metal vocals are achieved by some particular effect, but it really comes down to a vocalist who has their technique down and careful application of compression, eq, verb etc. Thanks. SM7b is a little out of my price range at the moment though. We're going to be finishing up the album in the next few weeks and I just don't have enough time to save money for that. Don't laugh, but how would an SM57 do? I guess we'll see! Heh.
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Apr 17, 2011 22:08
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I just got an AT2020USB microphone to make recording voiceovers and silly podcasts sound nicer. Unfortunately, I'm having a hell of a time monitoring it while recording. I'm new to using USB input microphones and was wondering if any of you had ever gotten a USB mic to monitor without noticeable latency? I'm running WinXP with 4GB RAM on a dual core processor, ASIO4All drivers with the samples down to 128, Adobe Audition 3 to record into and am listening through a Soundblaster Audigy Platinum ZS. Prior to this I had been using a cheap headset and never noticed any latency recording vocals directly through the Soundblaster.
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Apr 18, 2011 00:14
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BBW FEVER posted:Thanks. They're very, very similar mics - the 7B has a nicer transformer and a better housing, but they're incredibly similar. http://www.gearslutz.com/board/low-end-theory/470100-difference-between-sm7b-sm57-sm58.html
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Apr 18, 2011 15:51
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CuddleChunks posted:I just got an AT2020USB microphone to make recording voiceovers and silly podcasts sound nicer. After further fiddling with the ASIO drivers and Adobe Audition, the latency is tolerable. It's gone from a seriously irritating delay to barely noticeable. Now it just sounds like I've got a chorus mode on all the time set to "Totally Awesome" rather than an awful echo chamber. I can multitrack well enough for what I do I'm going to call this problem solved for now.
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Apr 19, 2011 06:34
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Carl Pagan posted:What are my options for mixing youtube videos when they are converted into Mp3's? I am trying to make a crisp version of this video for a friend. One is to have your audio in your original upload as uncompressed 24-bit PCM audio. It helps to have your original movie uploaded as a 1080p h.264 video as well. Even if you have to resize 480i/p footage to fit a 1080 line height. Also, watch the peak level when you upload. I'd go with SMPTE standards and have your audio sit around a level of -20 dB RMS. It amounts to a large upload but you really do end up with the best audio quality. This video was treated the exact same way I'm talking about... http://www.youtube.com/watch?v=bNOVqaDREos Edit: I take it back, this video was uploaded around -10 dB RMS with peaks at -0.2 dB FS. Would probably sound better with peaks ~ -3 dB FS. Hogscraper fucked around with this message at 03:47 on Apr 20, 2011 |
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Apr 20, 2011 03:43
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Hogscraper posted:Normalization is a bad practice. Don't mix or EQ for Youtube. Mix and EQ to sound good. There are some tricks you can do to get better quality audio on youtube. Ok great, I'll give these things a try and report back, thanks!
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Apr 20, 2011 16:31
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Got two Q's: -I'm running PTLE8 with an Mbox 2 on a C2D 2007 iMac. It's poo poo. Is it any use to go and get PT9? -I'm using a Clavia Nord rack as a bass synth. What should I try to make it sound a bit rougher/vintage/you know what I mean ? Stick a Bass Pod XT in there, or go with some sort of plugins?
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Apr 23, 2011 14:02
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Can any body give me a small beginnners tutorial on how to record using audacity? I have a sm57 mic. I want to record guitar tracks and then solo over them. I have been using grados headphones to listen to myself while recording, and using a click track to stay on track. However, its still not really sounding good yet. There is a lot of static noise in the background and its not picking up the guitar very well, it sounds really really different from what I am playing live.
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Apr 24, 2011 20:22
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What are you using to get the sound from the SM57 into your computer? That's probably the source of your problem. As for recording 2 tracks with audacity, it's really simple. Record one track, create a new track and record into that one. Then you'll have to do a little fiddling with the second track to make it align with the first.
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Apr 25, 2011 00:05
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I've got an MAudio Keystation and a Maschine that I'd love to put together on one big keyboard stand. Does anyone have any recommendations?
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Apr 25, 2011 22:01
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Just a quick question. Tomorrow I have to quickly record a few acoustic guitar tracks and we have two mics available: a small diaphragm condenser and a vocal dynamic mic (SM58). I know the main recording techniques for an acoustic guitar but I was wondering which one would be better considering the two different mic types. As far as I've read the XY technique really works better when both mics are the same line/model. Any help is greatly appreciated
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Apr 27, 2011 16:21
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How can I make a poo poo sandwich taste as little like poo poo as possible? Let me rephrase: I have a ghetto home recording setup and I have two sets of computer speakers neither of which are proper reference monitors. I've debated getting a decent pair of monitors, but it seems like from everything I've read that the cheap monitors aren't really worth the effort, so I think my best bet is to choose between these two sets I have and then listen to music on them until I get use to them, learn their weaknesses, and then mix with them (and of course try my mix on as many other speakers as I can). For the record I'm using an E-mu 1212m PCI card with Win7 and Sonar. The two systems I have are Altec Lansing ACS295 (http://support.dell.com/support/edocs/acc/ACS295/Specs.htm) and Logitech X-350 with only the front left, front right, and sub hooked up (http://www.logitech.com/en-roeu/speakers-audio/home-pc-speakers/devices/211). The Altec's have a slightly wider freq response and the left and right speakers are 2-way. The Logitech have more power, a better SNR, and each satellite has two speakers cones of the same size and I believe frequency bandwidth. While both have a knob for bass on the sub, the Altec's have a 2-band (T and B) EQ whereas the Logitech have none. My music style varies, but I do mostly electronic and do care about good low-end. So does any of this really matter or since they are both consumer computer speakers it really comes down to which I like the sound of better? Edit: For what its worth I can remove the grills from the Altecs which seems to be a good thing considering the covers have a plastic waffle in front of them. fallenturtle fucked around with this message at 22:45 on Apr 30, 2011 |
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Apr 30, 2011 22:23
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rt4 posted:What are you using to get the sound from the SM57 into your computer? That's probably the source of your problem. I am using a Cable to Line-in adapter, I think the technical term is like a 1/4in to 1/8in connector. I'm basically plugging into the line in on my computer. Does it really matter what sound card my computer has? Will I get a higher quality end product if I use a better sound card, or am I better off concentrating on my guitar, amplifier, microphone, and effects. By the way, I have been playing around with lining up tracks in Audacity and I got my input lag almost smack dab down, now that I can line up with the click track it's sounding wayyy better. But still not something I would want other people to hear yet. And that's what I'm going for. Any assistance is much appreciated!
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May 2, 2011 05:04
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Anybody know whats a good place to find wierd old connectors? I've just purchased myself off ebay an old dbx 158 chassis with 8x dbx 140's to try and tame a bit of the tape hiss off my Tascam 38 reel unit (It could probably do with a bit of surgery to replace some of the old caps, but $$$ and all that) but god drat if it doesn't have some wierd rear end plugs in the back. In theory I might be able to get at the little RCA's inside it (I believe these ought work) but it'd be nice to actually plug this sucker in as god intended it to. But gently caress me if these connectors aint wierd as poo poo. Their sort of like scart plugs, sort of , but with this wierd little pole I guess designed to hold it all still. I have no idea as to how to identify the fuckers and find some replacements so I can wire up the cables. Wierd fucken things.
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May 2, 2011 05:18
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fallenturtle posted:I've debated getting a decent pair of monitors, but it seems like from everything I've read that the cheap monitors aren't really worth the effort, so I think my best bet is to choose between these two sets I have and then listen to music on them until I get use to them, learn their weaknesses, and then mix with them (and of course try my mix on as many other speakers as I can). Even a cheap lovely set of behringers are going to sound more accurate than computer soundcard hifi speakers. The thing is , your not after "good" sounding, your after "accurate" sounding, and from there you just need to really get your ears used to the fucken things. The problem with home stereo speakers is they are tuned to sound cool by hyping up the bass and and poo poo, but they just are not accurate. In theory if you can get your mix to sound good on some proper monitors then it OUGHT sound decent on home stereo speakers, but that doesn't necessarily work the other way around, because your missing a lot of detail and mixing to a set of speakers that really just encoding to the assumptions of whatever company built the hifi speakers thought punters would enjoy. In the case of computer speakers, this is going to include explosions and guns firing and characters yelling "ROGER CAPTAIN" and poo poo that has nothing to do with music. If you can scratch the cash together for them, I honestly recomend some KRK's though. They really do pack a lot of punch for what you pay for them. Ultimately you want to just get some REAL monitors, even some cheap ones, and then listen to a lot of music through them to get your ears used to what a good mix should sound like on them (try and find music thats got decent dynamics, but whatever) and then aim at getting your mixes to sound like that.
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May 2, 2011 05:32
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coolbian57 posted:I am using a Cable to Line-in adapter, I think the technical term is like a 1/4in to 1/8in connector. I'm basically plugging into the line in on my computer. Does it really matter what sound card my computer has? Will I get a higher quality end product if I use a better sound card, or am I better off concentrating on my guitar, amplifier, microphone, and effects. By the way, I have been playing around with lining up tracks in Audacity and I got my input lag almost smack dab down, now that I can line up with the click track it's sounding wayyy better. But still not something I would want other people to hear yet. And that's what I'm going for. Any assistance is much appreciated! Yeah, sound cards on computers usually are passable for output but tend to be bloody horrible for input. Theres a few M-Audio type units that are pretty drat cheap and have pre's on them that aren't great but aren't terrible either. Your soundcard inputs are designed for little dinky headphone mics for using skype or whatever. You want something better than that. 57s are great little instrument mics, your doing it no favor by making GBS threads up their qualities with a generic soundcard input.
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May 2, 2011 05:37
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duck monster posted:But gently caress me if these connectors aint wierd as poo poo. Their sort of like scart plugs, sort of , but with this wierd little pole I guess designed to hold it all still. http://cgi.ebay.com/ADAT-Male-Female-XLR-ADX-164-Snake-Cable-4M-ELCO-EDAC-/190527404929?pt=LH_DefaultDomain_0&hash=item2c5c517b81
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May 2, 2011 06:28
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