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Roctor posted:So I am thoroughly a recording novice, and only do it as a hobby and to save stuff that I do. SHORT ANSWER: When recording, turn up your preamp and arm your track, then audition the track by playing your instrument or singing or whatever, while watching the meter on your computer screen. Make it so that you just barely won't ever clip the track on your computer's DAW (digital audio workstation)(in your case, Ableton Live). After that, if your source is still too quiet to your ears, you would need to apply some compression, which makes the quiet parts louder, and makes the loud parts quiet. As a result, you can turn up the whole track at the expense of dynamics (the relative difference between the quiet and the loud is now reduced). There are free compression VST plugins available everywhere; this compression is best applied after you have recorded your source. ______________________ Stuff like classical music rarely gets compressed, somewhat because the difference between pianissimo and fortissimo is supposed to be so great and in order to capture the performance as you'd hear it live, that difference has to be maintained, and also because there is a relatively high noise floor when a bunch of people are sitting in a big room together playing instruments, and you are recording from thirty feet away or whatever. Stuff like modern radio-ready pop gets compressed so much that it's called a "brickwall limiter" which is a super-form of compression, in order to have these ridiculous tracks almost always using the maximum amplitude available at a particular volume level on the speakers of whatever playback device you are using. In this way it cuts through noise very well, but also is very fatiguing on the ears after a time. ______________________ LONG AS HELL ANSWER: OK so there are a few terms you need to know and I'll probably flub some of this so wait for RivensBitch or wixard or Hogscraper or someone else to correct me, but it's more or less a layman's understanding: dBVU = the little needles which indicate "loudness" on old analog mixers, and sometimes the LEDs on newer mixers which "simulate" the little needles in terms of "loudness". The VU means "volume unit", and of course a dB is decibel, which is a (logarithmic) unit of measurement which indicates the intensity of a sound. For more read the last paragraphs here. Once you go over 0 dBVU, in an analog setting you'll get a nice analog distortion (if you've got tube preamps, and maybe sometimes if you've got solid state and you've tried it out and they sound good, although this is like guitar amps in that tubes distort "sweetly" and transistors (solid-state) distort "harshly"). On magnetic tape, you also would get a natural compression. Doesn't that all sound nice? On a vinyl record though, you'd get the needle skipping out of the groove (more or less). dBVU is in reference to your preamps, which may be on a mixing board, or may be their own separate deal. dBFS = the little color bars which indicate "loudness" on the screen of your DAW. The FS stands for "full sweep", implying a maximum amplitude wave. In a DAW, the absolute maximum is 0 dBFS. Above this, you get digital clipping which more or less sounds like a needle scaler just brushing against some metallic tweeters, or like your lovely cans breaking under the strain of the garbage music you are recording. There is some room for error on this - consumer gear almost always has a fudge factor built in and it differs from manufacturer to manufacturer as well as between specific products - but for our purposes, 0 dBFS is our maximum, it's when the little squiggly-line waveforms hit the ceiling and floor of the little bar they inhabit and go from describing neat little parabolas to describing ugly, nasty squares. dBFS is always in reference to the little colored bar on your DAW's screen. Noise floor = in layman's terms, this is the lowest possible level at which you could record and still have a retrievable sound play back, below which all you'll get back is the hiss of electrical interference. In amateur gear, this number is relatively higher than in pro-audio gear, but way, way lower than it was on your boombox when you were a kid and recorded your farts onto a cassette tape. As well, the more tracks you introduce, the higher the noise floor, since all that background hiss is additive. In general and unless you are Aphex Twin or something, noise floor is something to avoid, since it limits your options. Something else to keep in mind: once you've recorded to digital medium, you aren't going to introduce more noise into a particular track than is already on that track. What I mean is that the "faders" are digital and do not function the same way as your mic preamps, they have no native noise to add to the track. There will still be a noise floor, and a track will get "hissier" as you turn it up, but that is the "hiss" that was already there. You aren't adding more by turning up your tracks. Because of this, your goal is to record source tracks with the lowest proportion of noise floor, no matter what that level might be through your system, because once you've got the source recorded, you can turn it up as loud as you want in your DAW, and you aren't going to introduce more noise. This is only if you are trying to get the loudest source possible, but that usually isn't the main goal. What you really want is the best-sounding source possible. ______________________ There are two goals w/r/t recording sources in a digital medium. The first and foremost to keep in mind is to optimize the sound you are getting out of the preamps you are using to drive your microphone. There is a "sweet spot" in preamps, depending on what you want to do. Regular Joes like you or me start out by recording a source with an average dBVU of 0, with maybe some peak transients (meaning an oddly intense pluck of a string, an accidental drumstick mis-hit or what-not) up at +3 or +6 or so dBVU. A good thing to remember is that those numbers are logarithmic so +6 is actually quite a bit "louder" than 0 in this case. Are the bars on the track of your DAW clipping? That's above 0 dBFS then, and that's bad. If it's so, turn down 3 dBVU or so on your preamp, and try again. If not, go ahead and record whatever you like, then listen back to what you recorded. Next, try turning up your preamp a bit, and giving it another listen, and turning it down a bit and listening again. It's important to remember that if you turn up your preamp 3dB, to turn down your DAW 3dB before you play it back to listen, and if you turn your preamp down 3dB, turn your DAW up 3dB. The reason for this is that you don't want the guitar or whatever to sound "better" in one track than another simply because it is "louder" (a common psychoacoustic predeliction among us amateurs). Try this at a few different settings. If you do not have a dBVU meter anywhere in your signal chain, depend on your DAWs dBFS meter. Start at an average of -18 dBFS and go up from there in 3 or 5 or 6 or whatever dBFS increments. Now compare how each track sounded. Again, make sure they are normalized one to the other, and the guitar is as loud in each track on your dBFS meterse. In general, you should get the best-sounding source somewhere around 0 to +6 average dBVU (some above, some below) on your preamp, or -18 to maybe -8 average dBFS (some above, some below BUT NEVER CLIPPING) on your DAW. ______________________ Another school of thought says "USE ALL THE BITS" which is sort of a 16-bit way of thinking about things. This school of thought said to turn up your signal to the computer - through your preamp or mixer - by any means necessary, in order to get it so that you were peaking right at 0dBFS. In the 16-bit realm, this kind-of made sense, because there were only so many bits to go around, and if you added up a bunch of tracks there was a lot more tendency for the noise floor to go up. We're not in the 16-bit realm anymore, at least not in our DAWs. We're at the 24, 32 or even 64 bit stage - which some might claim is overkill, and others might claim is finally approaching the ability to mimick higher orders of timbral overtones: a place where a digital source becomes indiscernable to any human ear from a natural source. What I've noticed is that I'm turning down the trim on my DAW on every track - when I'm mixing a whole song down - something like -10 dB or so, or more. Now what this could mean is that I recorded every track too hot, since there is a bunch of signal I didn't actually use. I think wixard, however, is of the opinion that in general, with our amateur recorded gear, we can sort-of ignore the meters on our preamp source and just use the dBFS on the DAW, since that more or less will put us in our preamp's sweet spots... but it's important to know how all that stuff interacts in case you're interested in bending the rules a bit. Schlieren fucked around with this message at 07:50 on Aug 6, 2011 |
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Aug 6, 2011 07:29
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Man that is an awesome post.
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Aug 6, 2011 16:03
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What does the waveform look like in ableton live?
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Aug 7, 2011 01:48
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 edit: posted this without being 100% sure what waveform meant. googled it, pretty sure i'm a badass.
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Aug 7, 2011 05:07
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If the master and channel volumes are at 0 (default) then there is nothing wrong with that waveform, except it may be clipping in a few spots. I would turn the preamp gain down 3dB or so. If that is too quiet then you need to turn up your headphones/speakers. The only other issue might be one of routing your outputs correctly, or maybe there are plugins on the channel or master that are somehow making things quiet. If you master meters are green and at full level (as they should be with that waveform) then you're looking at an issue with your monitoring, not with your recording levels. On the matter of how loud to record, that waveform is just a bit past what I shoot for. I try to make the waveforms as big as possible so that I have tons of headroom when mixing, and so that I'm minimizing any noisefloor in the converters (which is not as big of a concern unless your preamps and converters aren't particularly good, but I don't leave anything to chance since it wont hurt to record a bigger waveform). Doing this has the added bonus of forcing me to bring all of my faders (in Ableton you use select all + drag fader, in protools you select the "all channels" fader group) down to -9dB or more, giving me a ton more room to adjust as I'm mixing. edit 2: VVVVVVVVVV this is bad advice, don't do this please. At the very least it's out of date as radio is less and less relevant every day so the loudness war doesn't matter nearly as much as it once did. RivensBitch fucked around with this message at 23:41 on Aug 8, 2011 |
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Aug 8, 2011 23:17
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Pretty much everything is too quiet before mastering due to loudness wars-era mastering techniques these days. I don't know of a good freeware VST mastering plugin, but you should probably find one and put it on your master bus.
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Aug 8, 2011 23:38
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Schlieren's post is pretty great - there are a few hiccups (DBFS = "full scale," not "full sweep," for example) but the general sentiment is spot on. There are two universal maxims that anyone doing digital recording should beat into their brains about gain staging: 1. 0 DBVU is equal to about -18 DBFS on most digital systems (because it's generally what their ADC's are calibrated for, because most analog outboard gear is designed to run best at around 0 dbVU). Record most of your tracks so they average around -18 on the digital meter, with peaks going as high as -6 or so on dynamic sources like drums. Don't be afraid to even track as low as -20 or lower. This is going to seem "low" and "quiet" and your resulting waveforms will be "little." BECAUSE 2. 24-bit recording pushes the noise floor down so low that we can actually record way quieter than we could in 16-bit without ever getting close to the noise floor. More bits = a deeper pool, not a higher cliff.
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Aug 9, 2011 04:55
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If it's too quiet, turn up your monitors or headphones, not the individual tracks. When you start doing proper gain staging, you'll find that your tracks just sound better (the ADC is working pretty much exactly at the level that it's been designed to, so the conversion of the sound from the analog domain to the digital sample is going to be the best possible version that it can be) and you'll even find that your tracks just plain start to mix together even better and will even start to "mix themselves" to a certain extent, especially with dynamic material like drums. You'll also find that as you're mixing many tracks, you're not overloading the master bus and having to turn down faders and pull down levels or mix into a limiter to avoid clipping - you have a ton of wiggle room to play with your levels and you'll find a real sense of air and space in your mixes.
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Aug 9, 2011 05:00
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RivensBitch posted:On the matter of how loud to record, that waveform is just a bit past what I shoot for. I try to make the waveforms as big as possible so that I have tons of headroom when mixing, and so that I'm minimizing any noisefloor in the converters (which is not as big of a concern unless your preamps and converters aren't particularly good, but I don't leave anything to chance since it wont hurt to record a bigger waveform). Doing this has the added bonus of forcing me to bring all of my faders (in Ableton you use select all + drag fader, in protools you select the "all channels" fader group) down to -9dB or more, giving me a ton more room to adjust as I'm mixing. With all due respect (the thread is named after you, after all!) waveform size is completely irrelevant. Proper gain staging and proper loading of the ADC converters is the thing that should determine the initial recording level. You refer to making the waveforms as big as possible in order to have more headroom when mixing, but this is actually the opposite of factual - a quieter signal that has been properly recorded around -18 on average (maybe -6 for peaks) is going to have more headroom to work with throughout the entire signal chain than a "bigger" waveform that's overdriven the ADC on the way in in the first place and already compromised the signal quality before you touch a fader. It's definitely good practice to throw all faders down before mixing, but you'll find that with proper gain staging like I described above, you won't even have to do much fader lowering in the first place and will find that your signals already sit nicely. If the waveform's too small, (almost?) every DAW has a way to zoom or magnify the waveform to taste.
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Aug 9, 2011 05:05
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Transistor Rhythm posted:If the waveform's too small, (almost?) every DAW has a way to zoom or magnify the waveform to taste. The DAW I use was coded in 2002 and has a zoom function. If yours doesn't you should probably upgrade. Transistor Rhythm posted:When you start doing proper gain staging, you'll find that your tracks just sound better (the ADC is working pretty much exactly at the level that it's been designed to, so the conversion of the sound from the analog domain to the digital sample is going to be the best possible version that it can be) and you'll even find that your tracks just plain start to mix together even better and will even start to "mix themselves" to a certain extent, especially with dynamic material like drums. You'll also find that as you're mixing many tracks, you're not overloading the master bus and having to turn down faders and pull down levels or mix into a limiter to avoid clipping - you have a ton of wiggle room to play with your levels and you'll find a real sense of air and space in your mixes. This seems super counter-intuitive and only makes sense upon practice.
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Aug 9, 2011 10:12
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Schlieren posted:This seems super counter-intuitive and only makes sense upon practice. Yeah, it does - especially to anyone who got started with 16-bit recording and poo poo-rear end converters back in the nineties, when the common knowledge was to "use all teh bits!"
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Aug 9, 2011 16:53
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Transistor Rhythm posted:..than a "bigger" waveform that's overdriven the ADC on the way in in the first place... If the over LEDs on my Rosetta 800 are flashing, then I'm distorting the waveform visibly in the DAW. Getting a full, large waveform in the DAW with peaks close to but not reaching 0, without any clipping on the converter, is easy to accomplish. So while I appreciate that you're using hard numbers here, my experience contradicts what you're saying. I get the argument that the size of the waveform itself is dependent on how the daw is displaying it, but most DAWs are set default so that clipping is indicated when the waveform is touching the vertical boundaries of the track. So using that as a reference isn't a bad idea. I also get the argument that with 24bits we're arguably way past the dynamic range of the analog noise floor, and that there's plenty of wiggle room. I would still argue that there is no downside in maximizing your dynamic range by getting as strong of a signal as possible without clipping, and I prefer to err on the side of more dynamic range, not less. If you were to argue that a lot of people in this thread might not have an ADC that is separate from their preamp/interface and that they can't meter the actual ADC stage to watch for clipping, I might concede that point, except that in practice I've never hit the peak/over meters on my converter when I use the size of the waveform as a reference. If you'd like to explain to me why I'm wrong, I'd love to hear it, but please don't try to tell me that I'm overdriving something when I'm not (I've recorded a lot of pure sine and square waves in my day, I can tell when something is being recorded correctly). And finally, I can't tell you how many digital mix sessions have come my way (it's in the hundreds) where all of the waveforms were really small and all of the faders were maxed out. It's quite possibly the biggest, most common mistake I see coming from other engineers; it's so prevalent that my most common move when taking over a session is to select all the audio regions and boost by 6-10db, and then reset all faders to -12db. RivensBitch fucked around with this message at 17:18 on Aug 9, 2011 |
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Aug 9, 2011 17:11
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Schlieren posted:I think wixard, however, is of the opinion that in general, with our amateur recorded gear, we can sort-of ignore the meters on our preamp source and just use the dBFS on the DAW, since that more or less will put us in our preamp's sweet spots... but it's important to know how all that stuff interacts in case you're interested in bending the rules a bit. Where I tend to disagree is when people come up with a number (like -18) that they have decided is optimum by looking at spec sheets and doing math based on 24bit noisefloors. Because in practice I'm pretty sure my $1200 interface doesn't have more than 16bit worth of headroom before you get to the noisefloor of its preamps with 30dB of gain. Transistor Rhythm posted:You refer to making the waveforms as big as possible in order to have more headroom when mixing, but this is actually the opposite of factual - a quieter signal that has been properly recorded around -18 on average (maybe -6 for peaks) is going to have more headroom to work with throughout the entire signal chain than a "bigger" waveform that's overdriven the ADC on the way in in the first place and already compromised the signal quality before you touch a fader. I think when RB says you have more headroom he means all of your tracks are as far as possible from the noisefloor. If session 1 is recorded at -12dBFS and session 2 is recorded at -22dBFS, you can trim/gain the tracks to set your faders wherever you want without effecting the resolution or noisefloor. No matter what you do with trim/gain plug-ins, when you mix them and start piling on compressors you will reach the noisefloor of session 2 before you will reach it in session 1. Maybe you won't notice a noisefloor in either session, that would be rad. But there is a logical reason to maximize your input. quote:It's definitely good practice to throw all faders down before mixing, but you'll find that with proper gain staging like I described above, you won't even have to do much fader lowering in the first place and will find that your signals already sit nicely. ChristsDickWorship fucked around with this message at 18:03 on Aug 9, 2011 |
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Aug 9, 2011 17:51
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Wax On posted:I can't see what you bought because the link isn't working but I had an M-Audio FastTrack for six years and never got it to work properly. The line input was never consistent and would go from a decent clear guitar signal to tinny crud without any notice. Once I upgraded to Windows 7 64-bit it would stop working randomly and I'd get ASIO sample rate errors from everything, even with current drivers. It seems a lot of other people had the same issue with their drivers on their current FastTrack as well. Yeah...it looks like they've evolved, but I bought this piece of poo poo: http://www.amazon.com/M-Audio-US44010-Interface-GT-Player-Software/dp/B00061ZM2Y I want it to die in a smelly dumpster full of ironic penises and neon plastic shades, somewhere in Silverlake or Echo Park. Maybe while listening to solo female singer/songwriter acoustic rock from the mid-90s. Now, I only paid $50 for it new thanks to stoned GC employees, but everything that every comes in through it sounds like I'm just using my mouth to make air guitar noises through a pillow. I loving hate this thing. In a more positive note, my progress with the Digi002 is coming along nicely.
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Aug 13, 2011 01:12
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I decided to look through DAWs while my recording equipment is in the mail (a Pod HD and a cheap dynamic mic that I probably won't be able to break), and the sheer number and diversity of all of them loving blows my mind, especially since I also have a MIDI keyboard hooked up that I planned on using with Sibelius 6 to fill in anything that isn't my voice or a guitar/bass. So far, assuming I don't pull up some speakers and a tape recorder which is getting much more tempting, REAPER seems to be my best bet, seeing it's within my price range (and the other DAWs within the price range like FL Studio seem to be catered more towards loops or electronic music instead of multitrack recording) and supports ReWire so I can sync it up with Sibelius easily (and in theory transfer audio, but while the sync works I can't get Reaper to actually record any goddamn audio from Sibelius), but I haven't seen it mentioned anywhere around here. Is it something I should be avoiding, or alternately is there something else I should be checking out that might be more to my liking?
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Aug 15, 2011 08:06
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Whatever you get, make sure it supports sidechaining. I just spent 6 hours of a Sunday creating a 450 point envelope to control my EQ to duck my bass out when the kick activates (luckily I a/b'd it and it actually works).
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Aug 15, 2011 09:07
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RivensBitch posted:If the over LEDs on my Rosetta 800 are flashing, then I'm distorting the waveform visibly in the DAW. Getting a full, large waveform in the DAW with peaks close to but not reaching 0, without any clipping on the converter, is easy to accomplish. So while I appreciate that you're using hard numbers here, my experience contradicts what you're saying. I specifically used the term "overdriving" the ADC rather than clipping, because they're definitely two different things. Even "overdriving" is a bit of a misnomer, but it's better than "hitting the ADC with a hotter signal than it's calibrated for, thus compromising the recording quality when you record a lot of too hot tracks that are all going to stack up in a mix" or something similar. Most ADC's are calibrated so that they operate best right around 0 dbVU - that is, they're going to do the best job of converting that analog signal to its accurate digital counterpart, with fewer quantization errors, jitter, non-linearity events and other things that subtly compromise the quality of the resulting digital signal - when the incoming analog signal is right around 0 dbVU. Different ADC's are calibrated differently, but in general, a majority of them are calibrated to -18dbfs = 0 dbVU, give or take. A few are more conservative (-20) and some older ones were calibrated hotter, around -12. The ultimate authority is the manufacturer's spec sheet, but I can guarantee that there aren't any that are calibrated to -6 or -3 = 0 dbVU...so when you track that high in order to "use all the bits" or "get a louder signal" or "get a big waveform," you're generally going to be hitting the ADC significantly hotter than it's intended to be run, and you will ultimately compromise your signal quality. It's not often going to be an obvious "distortion" - we're talking about near-audiophile/engineer nerd subtleties - but it will have an impact on the quality of your output by the time you've repeated this subtly destructive process over many recorded tracks. The best thing you can do when recording digitally is to learn this proper gain-staging, especially on cheaper equipment that uses cheaper ADC's (anything under $5K - haw!) that have far less tolerance for being run too hot and are going to subtly hash up your signals. Your mixes will thank you. Transistor Rhythm fucked around with this message at 17:29 on Aug 15, 2011 |
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Aug 15, 2011 17:20
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Schlieren posted:I think wixard, however, is of the opinion that in general, with our amateur recorded gear, we can sort-of ignore the meters on our preamp source and just use the dBFS on the DAW, since that more or less will put us in our preamp's sweet spots... but it's important to know how all that stuff interacts in case you're interested in bending the rules a bit. And that's definitely going to be true, in general - just as long as people remember that the preamp's sweet spot is going to be much closer to -18 dbfs than -6 or -3 dbfs! The cheaper your gear, the happier your pre is going to be around -18 to -20.
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Aug 15, 2011 17:28
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Transistor Rhythm posted:Most ADC's are calibrated so that they operate best right around 0 dbVU - that is, they're going to do the best job of converting that analog signal to its accurate digital counterpart, with fewer quantization errors, jitter, non-linearity events and other things that subtly compromise the quality of the resulting digital signal - when the incoming analog signal is right around 0 dbVU. Now, converter spec sheets that are worth a poo poo do not say "0dBV" or "0dBu" anywhere on them, what they tell you is what input dBU relates to 0dBFS. Here is RME's ADI-8 technical specifications from the manual: RME posted:4.1 Analog Specs quote:It's not often going to be an obvious "distortion" - we're talking about near-audiophile/engineer nerd subtleties - but it will have an impact on the quality of your output by the time you've repeated this subtly destructive process over many recorded tracks. The best thing you can do when recording digitally is to learn this proper gain-staging, especially on cheaper equipment that uses cheaper ADC's (anything under $5K - haw!) that have far less tolerance for being run too hot and are going to subtly hash up your signals. Your mixes will thank you. By the way, since you mentioned jitter in passing: The most obvious symptom of jitter is noise floor. Bob Katz is the only person in the world I have ever seen claim to really be able to "hear" jitter, and I'm not sure I believe him (he claims he can tell the difference between a DAT and a CD based on jitter because he claims most CD duplicators cause jitter). If your internal clock sucks everything hisses, I've seen it happen with a 200ft digital snake so bad it sounded like pink noise (we clocked it to an Apogee Big Ben, it got pretty damned quiet). If you think your gear has an inferior clock (like most entry level interfaces), it's even more reason to keep your inputs as hot as possible, above the jitter noisefloor. Based on my experience, I think that poor clocking, poor grounding and poor shielding have the biggest effect on entry level interfaces and they all relate directly to noise floor. They are also all but invisible factors on a spec sheet. ChristsDickWorship fucked around with this message at 23:04 on Aug 16, 2011 |
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Aug 16, 2011 22:54
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I read some responses from searching earlier, but they were a bit light on the detail. Anyway, I'm looking at getting back into cheap pro-grade home recording and want a control surface with I/O on it. I'm interesed in the Dig 002 and 003, the 002 is about $650 or so used on eBay, and the 003 in the $750+ range. I like the 003's shuttle control better, and it looks like a better layout. Is there any surface/IO combinations that might be as decent with decent playback/shuttle controls? M-Powered/Protools equipment is not required, used is preferrable, and I'm looking to spend around $800 or so. I looked at the FW1884 a few years ago but was slightly disappointed for a reason I don't remember. Any surface with I/O or a good rack mount I/O and control surface someone is selling as a package is good too! edit: I might have just bought a OG Mbox for like $80, does Pro Tools 8 LE work on it? Is it decent enough to just add a control surface like the Behringer one and run with that?
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Aug 17, 2011 03:50
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DJ Commie posted:does Pro Tools 8 LE work on it? Probably. Though Pro-Tools 9 just came out and now no longer requires any kind of special hardware anymore. You can run it with any interface you want. Personally I think you're better off buying a separate audio interface and control surface, if only because you can tailor it to your exact needs, and buy or upgrade in pieces as needed.
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Aug 17, 2011 07:17
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As far as control surfaces are concerned, I'm currently developing a small (8" x 3" x 3/4") control surface that is a 40x2 character LCD screen with 8 channel and 1 master pushbutton encoder knobs, and a few modifier/shift buttons. It's arduino based and will emulate a Mackie Control so it's compatible with all major DAWs. My goal is to make it super compact and affordable.
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Aug 17, 2011 19:13
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RivensBitch posted:As far as control surfaces are concerned, I'm currently developing a small (8" x 3" x 3/4") control surface that is a 40x2 character LCD screen with 8 channel and 1 master pushbutton encoder knobs, and a few modifier/shift buttons. It's arduino based and will emulate a Mackie Control so it's compatible with all major DAWs. My goal is to make it super compact and affordable. Will buy this where do I send my monies.
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Aug 17, 2011 22:02
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I'm currently decoding the Mackie control protocol, in fact I'm building a max for live emulation of an MCU as a functional way of really understanding how the MCU works.  I have a pile of RGB illuminated pushbutton encoders that I'm evaluating to pick the right ones, so each knob will have full RGB color feedback to help you easily see what it is you're doing. Once I have the prototype working and the PCB designed I'll have to do some kind of presale as i've discovered that the RGB encoders are only available when you order direct in quantities of 10,000. Aiming to keep costs as low as possible, but still don't know what the final price will be. A Mackie Control costs $1000 and a behringer BCF2000 costs $175, I'm hoping to be cheaper than the behringer, if not assembled then definitely in kit form. RivensBitch fucked around with this message at 07:14 on Aug 18, 2011 |
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Aug 18, 2011 07:08
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That looks really cool, next step up I'll keep your project in mind, I am an electronic hobbyist also so that would be awesome to nerd out on! Today I scored a Tascam FW 1082 for $190 shipped, that fulfills all my requirements and leaves me some cash to buy Protools 9. Is it really hardware agnostic?
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Aug 19, 2011 04:50
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HotCanadianChick posted:Probably. Pro Tools 9 really doesn't care about hardware as long as you pay full price for it and buy an iLok.
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Aug 19, 2011 07:52
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RivensBitch posted:I'm currently decoding the Mackie control protocol, in fact I'm building a max for live emulation of an MCU as a functional way of really understanding how the MCU works. Will you be doing motorized faders?
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Aug 19, 2011 19:10
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The Electronaut posted:Will you be doing motorized faders? No. Motorized faders have already been done cheaply, check out Behringers BCF-2000. They also take up a bunch of space, they are what they are. There's really no point to reinvent that wheel. My controller will be SMALL, and instead of faders it will have encoders. But unlike the behringer there will still be an LCD to see the channel names and the values of the parameters you are changing (volume, pan, sends, plugin parameters, and a layer of MIDI CCs). Also, the encoders will each have an RGB LED inside of them that illuminates the shaft and the knob. In it's default setting the knobs will indicate fader position (brighter = more gain, or a higher fader position), but I'm also going to experiment with using one color for fader position, and one color for VU metering, with the third color for another parameter. This might seem less useful than a controller with motorized faders, but I think it has plenty of applications for computer based musicians who aren't locked into traditional analog mixing controls. Making the controller arduino based and open source also opens up the opportunity for expanding the uses of this controller, and I'm hoping that will have more appeal to electronic musicians than just trying to make a cheaper version of something that's already readily available. I also think the size is a huge factor, many electronic performers have crowded tables and the last thing they want to do is give up another 12"x12" of space just to make small volume adjustments. RivensBitch fucked around with this message at 22:40 on Aug 19, 2011 |
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Aug 19, 2011 22:37
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Oh! Ok, I saw you mention encoders but you mentioned the BCF-2000, thanks for clarifying. Very cool on you to do this and release it to the world when done. I wanted an Alpha Track a while ago but realized that just one fader wasn't enough.
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Aug 19, 2011 22:41
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RivensBitch posted:A Mackie Control costs $1000 and a behringer BCF2000 costs $175, I'm hoping to be cheaper than the behringer, if not assembled then definitely in kit form. Use mechanical switches (Cherry) for the transport buttons and I will order one. I'm tired of the rubber/tinny clicky ones.
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Aug 20, 2011 00:23
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(click images for full size)   The colors of the encoders as pictured are meant to signify red meaning the fader position (brighter red = a higher/louder fader), the green encoder is the currently selected channel (intensity still signifying fader position, just in a different color), and the blue master encoder could be blinking at the tempo, or perhaps blue could signify which "page" we're on (volume, pan, sends, MIDI CCs, etc). Other actions could provide feedbacj, like soloing a channel could darken all the encoder knobs except for the soloed channel which would be blue. Colors could also be animated, perhaps green would make a better fader position and then blinking red would indicate a record armed channel. All of the encoders are pushbuttons as well. I'm still developing the interface and how users will interact with it, I like the idea of the non-encoder buttons on the left and right being color coded modifiers. For instance to be in line with the MCU, make them blue, red, yellow, and green for solo, record, mute, and select. Hold down the blue button and press the encoder of the channel you want to solo, hold down the red to record arm, etc. But then the master encoder changes what page you're on, you can use it to flip to pans, sends etc, and then the color coded modifiers would do something else. Use the modifiers when adjusting the master encoder and that's how you reconfigure the controller. And of course using one of the right modifiers would turn the four left modifiers into banking buttons, by single channels or groups of 8. And when holding down a modifier, the LCD would temporarily blank the channels and tell you what that modifier does, so that way you don't have to memorize the whole thing. There's a ton of different ways to do it and I need to build the prototype so I can start using it and see what makes sense, and then make design changes based on that. Laserjet 4P posted:Use mechanical switches (Cherry) for the transport buttons and I will order one. I'm tired of the rubber/tinny clicky ones. I'm torn on this due to the size limitations and keeping costs low, could you give some examples of the kinds of buttons you'd like to see for transport? I could see using the design above and holding down one of the modifiers on the right to turn the modifiers on the left into transport controls, but my current intention is to make the buttons simple tactile switches with caps. I'm open to hearing suggestions though. RivensBitch fucked around with this message at 00:51 on Aug 20, 2011 |
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Aug 20, 2011 00:38
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recon_etc posted:I decided to look through DAWs while my recording equipment is in the mail (a Pod HD and a cheap dynamic mic that I probably won't be able to break), and the sheer number and diversity of all of them loving blows my mind, especially since I also have a MIDI keyboard hooked up that I planned on using with Sibelius 6 to fill in anything that isn't my voice or a guitar/bass. I was a long-time Cubase user, and I've now entirely switched over to REAPER. It was a little sketchy back in the 1.x days, but now it's definitely become a real option as a DAW (and to go along with it, they've finally come up with a default theme that's unique and professional-looking, which will do more to encourage people to give it a try than any feature set). Probably the most impressive things about it are how efficiently it handles multiple processors, and the constant progression of features and bug-fixes - I find myself just trying to do things I expect my DAW to be capable of, and if REAPER doesn't have it yet I usually find it in a changelog a month later. It's like they read my mind!
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Aug 30, 2011 05:19
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I've got a problem pretty much exactly like Roctor's, but I'm 100% new to recording (my stuff only came just the other day), so I tried to read through the tips on this thread but failed to comprehend much at all of it. I get it at a very superficial level, but I was hoping for some tips in very laymans terms. Basically, I'm trying to record electric guitar (and later, will be attempting to record acoustic when my mic comes this week) through a Focusrite Saffire 6 USB into Ableton Live Lite 8 (just because it came with the interface; I've never used any DAWs before so I'm just trying to explore this one). So yeah, I'm getting the same sort of problem where I turn everything up/down to the point where nothing's clipping, but when I play it back on Ableton everything's really quiet; so quiet in fact that I can barely hear it. When I'm monitoring myself while I'm recording it though, I can hear everything at a great volume in my headphones in realtime. Is the basic message here just to turn up my headphones? I exported some crap I just played around with in Ableton, and it was all at a perfectly good volume as a wav file playing through some other media player. I just want to find out how to hear the playback through Ableton better, because it's hard especially when I want to track something else on top of what I've already done; the instrument ends up being much louder than what I've already previously recorded and makes it hard to sync up. Edit: Also, I was wondering if I could get some advice on the sort of headphones to get for monitoring etc. I heard that open back are better for mixing(?) but I'm concerned about whether or not some of that sound leaking out while I'm recording acoustic guitar into a mic will be picked up by the mic. Chroisman fucked around with this message at 17:54 on Sep 4, 2011 |
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Sep 4, 2011 05:54
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Does anyone use amplitube/guitar rig with an x-fi sound card? If so would they be kind enough to post their settings in asio4all and the programs. I'm absolutely bamboozled and have given up trying to work it out myself.
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Sep 4, 2011 17:51
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Chroisman posted:Also, I was wondering if I could get some advice on the sort of headphones to get for monitoring etc. I heard that open back are better for mixing(?) but I'm concerned about whether or not some of that sound leaking out while I'm recording acoustic guitar into a mic will be picked up by the mic. The general consensus is that open-back are better, but there are also some decent closed-back headphones out there. I love my Sony MDR-V6s, they're identical to the MDR-v7506s that get used in TV and studios everywhere except they're $30 cheaper because they don't have a gold-plated plug. They're also absolutely indestructible, I've had mine since high school, (some 10 years now) and beaten the crap out of them, and they still work great. They're closed-back so great for tracking; the trick for mixing with them is to remember that they're a little on the bright side, which is good for me because I tend to be too aggressive in pushing the highs anyway. Do not buy the MDR-V600s, completely different headphones.
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Sep 4, 2011 18:23
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Pretty sure 7506s are the replacement to V6s, V6s aren't made anymore. The reason they were 30-50 cheaper 10yrs ago was because they were the old model (I bought a pair then too), they are identical and all the broadcasters 12 years ago were wearing V6s. 90% of what you read about mixing on headphones is bullshit being regurgitated by people who read it somewhere else. Every engineer I know can mix on headphones. It causes more ear fatigue and you can miss things that are out of phase, but there's no reason mixes done on headphones are inferior to mixes on monitors if you know the headphones well. I personally think a 100-150 pair of headphones are a much better reference for me than any monitors under $1200 a pair. YMMV, like Mradyfist I've owned Sony V6s or 7506s since I was 14. I don't think any pros who have ever made a comment about headphones sucking were comparing them to any kind of affordable monitoring solutions. ChristsDickWorship fucked around with this message at 01:08 on Sep 5, 2011 |
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Sep 5, 2011 01:01
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I never knew that the 7506s were officially intended to replace the V6s, but Wikipedia agrees with you. Regardless of whether they're currently being produced, you can still get brand-new MDR-V6s from a few hundred stores online, and they're still about $30 cheaper than the 7506s. I also agree that unless you have a lot of cash to spend on a decent setup mixing on headphones is going to be a better solution, even with decent monitors ($1200+ range) you still need to at a minimum consider the room and probably do some treatment.
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Sep 5, 2011 06:53
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Mradyfist posted:I never knew that the 7506s were officially intended to replace the V6s, but Wikipedia agrees with you. Regardless of whether they're currently being produced, you can still get brand-new MDR-V6s from a few hundred stores online, and they're still about $30 cheaper than the 7506s. I also agree that unless you have a lot of cash to spend on a decent setup mixing on headphones is going to be a better solution, even with decent monitors ($1200+ range) you still need to at a minimum consider the room and probably do some treatment. edit- Also according to wikipedia my 14yo self might be remembering prices higher than they were but I do remember the sale, and it not being clear that the 7506s were the same thing so the catalog made it feel like it was the last chance to buy them (buying without the internet lol). I had those for 8 or 9 years before a dog ate the tiny wire that went to one of the ears (even then I managed to solder it for a while, but not well) and I pretty much only saw 7506s everywhere and have had them ever since. They sounded identical to me switching over. ChristsDickWorship fucked around with this message at 18:46 on Sep 5, 2011 |
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Sep 5, 2011 17:52
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Mine were probably bought a little over ten years ago, and while they do say "Studio Monitor" over the headband they also have a red "for DIGITAL" sticker on the right cup, while the current 7506s have a blue "Professional" sticker in the same spot. I've also heard rumors of Sony accidentally shipping some sets with one cup labeled MDR-V6 and the other labeled MDR-7506, but I've never seen one myself.
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Sep 5, 2011 18:29
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The shure srh440 smoke the 7506, and I've owned my 7506s for 16 years.
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Sep 5, 2011 23:01
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