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Quiz Show Scandal posted:So ProTools doesn't support VST, but Reason does? Is there a way to make the two "join forces" so that I could take advantage of all the free VST stuff? Maybe create a VST track in Reason, export it as a finished audio file, then pull the file into ProTools? No, Reason does NOT support VST's. That's not even what your quote said! quote:LE comes with Reason Adapted I think, which I'm pretty sure has a basic piano and rhodes patch, but it's a little more involved to use Reason than the average virtual instrument. virtual instrument != VST. VST's can be virtual instruments, but not all virtual instruments are VST's. VST is a specific type of plugin. Reason is a program. HOWEVER, Reason supports absolutely NO external plugins whatsoever - it's got its (notice correct use of the two forms of its  ) own set of samplers/synths/effects and it comes with a crapload of presets and samples, but it is not and does not support VST's.
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May 20, 2007 19:59
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Ok I bought one of these http://www.m-audio.com/products/en_us/Uno-main.html so I could use my cheap Yamaha keyboard(YPT-200). I have this extremely annoying problem that I can't fix, when I hit the key the computer will play the first note but after the first note everything is just garbled crap. I have tried updating drivers, using a buch of different programs(fruity loops, cubase) and I recently got a new PC and it still gives me the exact same problem. Its not the cable because they guy at the store tried it and he got it working fine. Im thinking it could be my keyboard, but its new and everything else about it works fine. Anyone have any ideas of how to solve this?
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May 20, 2007 21:18
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Reason only supports REWIRE which is a competing format to VSTs 
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May 21, 2007 00:21
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RivensBitch posted:Reason only supports REWIRE which is a competing format to VSTs That's not a plugin! Stop being intentionally confusing!
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May 21, 2007 01:33
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VST isn't a plugin, it's a standard for communication between audio programs, same as re-wire. The difference is in the common application of the two. If a plugin uses VST as it's form of communication, we call it a "VST-Plugin" the same way that if we have a soft synth that uses VST to communicate we call it a "VST-Instrument". But there are other soft synths that support re-wire as well as VST, and we can call those "Re-Wire Instruments" the same as we might call them "vst-instruments". In fact when I use reason from within protools I use re-wire, I can consider it a rewire-instrument because I may not use it's sequencing at all but I will use the built in sampler as a sound module for drums that I program within protools. When I do that I simply create a midi track and pipe it's output to the reason sampler, same as if it was an RTAS sampler. But this is all semantics really, the point is that all of these protocols are different ways of accomplishing the same things. They're all simply different methods of exchanging midi, audio, and time code.
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May 21, 2007 05:14
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RivensBitch posted:Technically you're right, but here's how I understand the (very important) difference - ReWire allows two applications to communicate with each other - both applications can operate as stand-alone apps as well. A VST-plugin or instrument is something that cannot run on its own - it needs a host program (Cubase, for instance) in order to work. You may be able to find contradictions to this, but 99% of the time when somebody is talking about either of these things, this is what they mean. edit: And Reason can only send audio out, right? It cannot receive via ReWire as far as I know.
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May 21, 2007 06:37
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re-wire is a standard of communication between applications vst is a standard of communication between applications while reason may or may not receive audio via re-wire, that doesn't mean it couldn't. Fruityloops will act as both a re-wire host and a re-wire instrument, meaning it can both send and receive audio via re-wire. Absynth is a stand alone synth application that uses VST to communicate to the host program. While cubase SEES it as a plugin, that's only because cubase is designed with VST plugins in mind. You could easily write a program that acted as a VST plugin but was actually a stand-alone audio recorder. You could have it send audio straight through or mute it or put other plugins on it or nothing. If you've ever worked with max/msp you know that you can use VST to communicate between stand alone max patches that might act as plugins or they might be ANYTHING else. Cubase and protools and reason are all programs that are designed to behave certain ways and interface with various protocols. The user interfaces they put up around those protocols make it seem like vst is only for "plugins" and re-wire is only for reason, but that doesn't mean those protocols are actually limited to those uses. A perfect example of this trickery is the FXPansion vst/rtas wrapper. It basically creates an rtas shell plugin that acceses your vst plugins and pipes the audio/midi through. It literally takes the vst plugin and compiles an rtas plugin around it, creating a new plugin that's basically a stand alone program with an RTAS interface that's running a VST plugin inside of it. RivensBitch fucked around with this message at 08:51 on May 21, 2007 |
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May 21, 2007 08:47
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RivensBitch posted:Oh. Cool.
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May 21, 2007 17:36
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Let me ask a much more refined version of my original question - I'm still a bit confused, so my apologies. With ProTools LE and an MBox 2, will I be able to get a Rhodes piano sound? And yes, I do have a MIDI keyboard if necessary. If the answer to the question is yes, how easy or not easy would this be?
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May 27, 2007 06:21
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Quiz Show Scandal posted:Let me ask a much more refined version of my original question - I'm still a bit confused, so my apologies. ProTools LE generally comes with Reason Adapted, which should have a Rhodes sound. You'll have to learn to use ReWire to get the programs to talk to each other - open ProTools LE, THEN open Reason Adapted, and make an instrument track in PTLE with Rewire selected as input or something like that... I'm rusty on the details but there should be detailed tutorials somewhere (so don't ask here - use Google.)
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May 27, 2007 06:51
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I would like it very much if some of our elder statesmen of recording would talk about EQing for a bit. I understand that the best records have niches carved out in the frequency spectrum for each instrument so that the mix sits tightly together and everything is happy, but I do not have the years of experience to know what freqs go where, beyond the obvious midrange jumble that male vox and guitars and pianos are fighting for. Thanks! Rock all up in your face like a sexy kiss.
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May 27, 2007 16:29
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ampleforth posted:I would like it very much if some of our elder statesmen of recording would talk about EQing for a bit. Another point of curiosity - will any sustain pedal that already works with my keyboard also work with MIDI applications? Can MIDI interpret a sustain pedal?
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May 27, 2007 17:59
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Quiz Show Scandal posted:Another point of curiosity - will any sustain pedal that already works with my keyboard also work with MIDI applications? Can MIDI interpret a sustain pedal?
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May 28, 2007 14:32
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Slight recap: right now I have a Mackie 24 VLZ Pro going into a Yamaha 01V with ADAT going into an E-mu 1212m. This is a clumsy setup, the 24 VLZ is too big, the 01V is only a converter, and from what I've read not such a great one in terms of sound, and the 1212m is decent but the sum of all parts isn't that hot. Earlier I was thinking about a new audio interface. I've got the shortlist down to: - RME Multiface + some ADAT box - MOTU 828 mkII Firewire + some ADAT box - Mackie Onyx 1620 I have mainly synths to hook up - don't care about mic preamps that much. I want to use my hardware FX with my hardware - so I'm not too concerned about the Onyx's 2 outs - unless there's a reason to be concerned which I don't see yet. I really like the idea that the 1620 will just work as mixer, too - call it longevity reassurance  Any ideas or am I simply approaching this the wrong way and is there a better solution?
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May 28, 2007 19:17
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ampleforth posted:I would like it very much if some of our elder statesmen of recording would talk about EQing for a bit. I understand that the best records have niches carved out in the frequency spectrum for each instrument so that the mix sits tightly together and everything is happy, but I do not have the years of experience to know what freqs go where, beyond the obvious midrange jumble that male vox and guitars and pianos are fighting for. Here's the secret. http://braincleaner.net/?q=node/9 For additive EQ, do the same thing, except when you find something that sounds NICE, widen the Q, then pull the gain down to 0, and slowly bring it up until you hit the sweet spot you like. Narrow Q when removing, wide Q when adding. PRADA SLUT fucked around with this message at 04:36 on May 29, 2007 |
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May 29, 2007 04:32
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YO I'm a DJ and I'm trying to get a kind of studio set up. Right now I'm using lovely computer speakers as monitors and it's not working for me, so I'm thinking of adding monitors. I'm also looking into an audio interface so I'll be able to record from my turntables to the computer, as well as output from my computer to the monitors for working on remixes/edits/etc. And finally I need a mixer of some sort that I can plug the output from the turntables and the soundcard into and run all of that to the monitors. For the monitors I'm looking at Behringer Truth Active, which run around $300. I know that Behringer is total poo poo but I've heard that these are the exception to that rule and you can't get anything better unless you spend several hundred more dollars. I'm a little stuck on the soundcard. I'm not going to be recording vocals or guitar or anything like that, so I don't really need good preamps. Like I said, I basically need to be able to record the output from my DJ mixer and be able to output from my computer, nothing too fancy. Any recommendations would be great for this. And for a mixer, I was looking at a used Mackie 1202 from eBay (there are several up there) for about $100 that seems like it'll do what I want. It seems kind of like overkill though, I just need to be able to hear the outputs from my DJ mixer and my computer through the same monitors without unplugging poo poo every time. Any help/advice would be great, I'm a little new to audio equipment!
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May 30, 2007 04:14
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ampleforth posted:... beyond the obvious midrange jumble that male vox and guitars and pianos are fighting for. It depends on the sources obviously, but I find that "mids" covers a lot of territory and those three instruments have their different spots. You have to use your ear of course, if you're using a plugin the with your mouse drag a greatly boosted width of frequencies across the spectrum slowly and listen to what you hear. it should become clear which frequencies matter and which dont, if you hear frequencies that sound bad pull them out, if you hear frequencies that are good leave them alone. Never boost. Take notes of the frequency bands that you pull out of certain instruments. If you are having problems making two instruments fit together in a mix, try to coordinate which frequencies you pull and which frequencies you leave in. Always remember that sometimes what you do to an instrument will make it sound good in the mix but might make it sound not so good on it's own.
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May 30, 2007 07:25
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ampleforth posted:I would like it very much if some of our elder statesmen of recording would talk about EQing for a bit. I understand that the best records have niches carved out in the frequency spectrum for each instrument so that the mix sits tightly together and everything is happy, but I do not have the years of experience to know what freqs go where, beyond the obvious midrange jumble that male vox and guitars and pianos are fighting for. While mixing our CD tonight I just discovered this page: http://www.espace-cubase.org/anglais/page.php?page=appaudiozg1 It definitely helped bring out the best in our horn section, but I haven't tried their settings with anything else. And granted, you're gonna have to play around a bit but the overall numbers given were a great starting point.
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May 31, 2007 06:06
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blingasaurus rex posted:Needing advice on monitors and interfaces and mixers and things and things. My laptop I am using has been great while I have been using software synths and samplers, but I recently got a hardware multi-timbral synthesizer and have been focusing on creating music on it, and only it. The problem is that my laptop does not have a line-in, it only has mic input. (Right now, my setup is really ghetto, it has 2 1/4" jacks coming out of the unit, it is converted to RCA, and from that to phono which goes into my laptop's mic port) My laptop does not have a line-in port. Because of this, I cannot record in stereo, and it's at the point that I really must have stereo. I don't need extras like XLR ports too bad because I think if I need that down the line I would just get a mixer and plug that into my interface. I would like to the option to be able to plug monitors later on if I decide to get some. However, I am not looking to spend more that $150 on an interface, maybe $200 if there is good reason. Pretty much the cheaper the better, but I know that you usually end up getting what you payed for the cheaper you go. I have looked around at what's out there, and I don't know if it would make a much of a difference to get something like a little Edirol UA-1EX (which is really barebones but is pretty much what I need) or something else with more features like direct monitoring. (would the UA-1EX technically have something like that?) I know nothing, and I would like some advice.
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Jun 1, 2007 07:56
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Hey guys. I'm trying to find a manual for a motu 2408 mk2. Any thoughts?
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Jun 1, 2007 19:06
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Right now I am trying to mix horns for my band's EP and they just do not sound right. The trumpet sounds too clean; how would I bring out the brassiness? (I know two posts up I posted a guide myself but further listening shows that it is still not entirely right). Also, any guides for panning? I have 2 tracks of trumpet and 2 tracks of trombone but I can't get them to sit quite right.
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Jun 1, 2007 20:06
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hedgecore posted:Right now I am trying to mix horns for my band's EP and they just do not sound right. The trumpet sounds too clean; how would I bring out the brassiness? (I know two posts up I posted a guide myself but further listening shows that it is still not entirely right). Also, any guides for panning? I have 2 tracks of trumpet and 2 tracks of trombone but I can't get them to sit quite right. If you recorded it with poor equipment, you won't be able to 'bring out' the brassy detail if it's not there to begin with. Sweep the EQ, like the guide says. That's pretty much all you can do. If you find what you're looking for then you've got it... if you can't find it, your recording sucks and there's nothing you can do about it. As for panning, just listen to your favorite albums. There are tons of ways to do it. I'd suggest trumpets moderate left and right and trombones more subtle left and right - that should spread it out, provided they're all playing the same rhythm most of the time. Also, use compression.
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Jun 2, 2007 18:25
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I'm thinking of making some DIY acoustic treatment for my apartment, nothing fancy as I don't really have the funds for it. Probably just insulation and the like. Any thoughts on inexpensive materials or construction tips? The only thing is that it all needs to be removable, since I don't own the apartment and will be moving sooner or later. Also ideas that don't leave an excessive amount of holes in the wall would be great. Right now I'm basically thinking of thin sheets of plywood with a thick layer of insulation and covered in some kind of cloth material to keep it all in and look pretty. Not sure about attaching it to the walls/ceiling though. Would this probably work? Edit: On re-examination, this question doesn't really belong in this thread. But, I guess I'll leave it just in case someone wants to answer
Elder fucked around with this message at 18:25 on Jun 3, 2007 |
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Jun 3, 2007 18:17
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Elder posted:I'm thinking of making some DIY acoustic treatment for my apartment, nothing fancy as I don't really have the funds for it. Probably just insulation and the like. Any thoughts on inexpensive materials or construction tips? The only thing is that it all needs to be removable, since I don't own the apartment and will be moving sooner or later. Also ideas that don't leave an excessive amount of holes in the wall would be great. Here's a guide on acoustic treatment: http://www.sweetwater.com/shop/studio/acoustic-treatment/buying-guide.php (You can ignore the sales pitches and concentrate on their explanations)
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Jun 3, 2007 19:39
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nimper posted:Do you want soundproofing or are you interested in using your apartment as a recording space? If the latter then you might want to re-think your plan. A little of both, but mostly soundproofing. I'll be recording mostly vocals, so having a nice, resonant space isn't that important. Basically I want a space where I can 1. Sing as loudly as I need without pissing off the neighbors 2. Mix properly If I can do this without dropping the cash for the professional acoustic treatment it would be great. Of course I understand that it will be far from ideal, but I figure that something is better than nothing. Anyways, it's just something I've been considering. If anyone has tried this, I would love to hear about the process and results. And that's a very good guide, thanks for the link. I have an understanding of the basics but it's been a little bit so it's a good refresher.
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Jun 4, 2007 03:21
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RivensBitch posted:It depends on the sources obviously, but I find that "mids" covers a lot of territory and those three instruments have their different spots. You have to use your ear of course, if you're using a plugin the with your mouse drag a greatly boosted width of frequencies across the spectrum slowly and listen to what you hear. it should become clear which frequencies matter and which dont, if you hear frequencies that sound bad pull them out, if you hear frequencies that are good leave them alone. Never boost. Take notes of the frequency bands that you pull out of certain instruments. If you are having problems making two instruments fit together in a mix, try to coordinate which frequencies you pull and which frequencies you leave in. Aye - I found vocals much easier to work with in the mixdown than say analogue synths and it comes down to the harmonic intervals. Pretty much every sound I have ever mixed with few notable exceptions (most 808 bassdrums with the attack setting low) span pretty much every audible frequency. Male vocalists will have their lowest harmonic interval in the lower mid ranges but the rest go way up to 11,000hz. And in between the (somewhat complicated) series of peaks you have alot of troughs. Whereas analogue synths for example output variations on basic waveforms. Harmonic sounds are often built out of saw waves and you could think of those as giant blocks of sound that carry a fundamental and all odd and even harmonics in linearly descending intensity down the length of the scale. So you shave off the stuff you don't want using a filter. And chugging guitar power chords are often tough to work with since the amount of distortion on them is typically high adding to the wall of harmonics that many hard rock bands try to create. Vocals are really different because the harmonics aren't so regular. One trick I use with paragraphic EQs is to switch the middle point on the EQ to a bell curve and narrow the Q so you get a very thin bell curve. Now turn down the master volume for the next bit. Increase the amplitue of the bell curve so its as high as it will go on the EQ. Sweep the entire range of the EQ to find the loudest point. Now increase the master volume so that its tolerable. This is just health and safety stuff. Now sweep the entire range of the EQ again with the narrow bell curve and listen out. You will notice the amplitude will suddenly spike at specific and ordered intervals along the length of the EQ. These are the harmonic intervals of that particular sound and are represented on a spectrum as a series of tall thin peaks in relation to a fundamental frequency (the lowest harmonic). With few exceptions there is mostly nothing below the fundamental (since this is the lowest frequency pitch reference anyway) so feel free to automatically hipass everything below the fundamental. Mostly when you layer laods of audiotracks of harmonically rich intervals then some ofthese harmonic intervals really overlap and start causing problems. Varying degrees of destructive phasing when summed to mono but mostly amplitude spikes caused by the frequency ranges of 2 different sounds being in the same phase. So it causes problems with headroom management. Do the trick I mentioned to find where the worst offending harmonic is. Once you have identified it (it will be the loudest) then you simply turn the bell curve upsde by giving it a negative gain. Voila - you now have a notch filter at the offending frequency. widen the Q as necessary. I tend to find wide Qs are better for where you want to subtle changes. If you hard notch out alot of harmonics you will notice that it can make vocals and a number of instruments liked violins sound very unnatural. And if you notch out the fundamental you can lose its pitch reference which can ound quite freaky. I tend to prefer using linear phase EQs for very sharp, very narrow notch/bell filters. Linear phase EQs are dead perfect and sterileand some of them sound it (Waves LinEQ for example). I tend to prefer non linear phase (analogue) EQs for very broad, shallow notch/bell filters. Analogue EQs will introduce a phase shift to the output signal and have other imperfections which some people find desirable (that are also mimiced by digital EQs that attempt to recreate the sound of analogue EQs). So yeah - digital EQs for precise surgical cuts and boosts. Analogue EQs for broad sweeping cuts/boosts.
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Jun 7, 2007 17:36
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WanderingKid posted:I tend to prefer using linear phase EQs for very sharp, very narrow notch/bell filters. Linear phase EQs are dead perfect and sterileand some of them sound it (Waves LinEQ for example). Did you make this up or did someone else actually create that bunch of disinformation? what makes you thing that digital EQs are inherently phase linear? the waves linear EQ is a resource hog because it's attempting to make up for phase smearing, which to me digital EQs are more prone to than analog. That said filtering is perhaps one of the tougher/expensive processes to create in a way that sounds pleasing to the ear. I don't think there's a clear dilineation between the mid range analog and digital EQs, although with analog you really have diminishing returns until you have a high quality console to connect everything (and those usually have top quality EQs on each channel). I would say that once you get into the high end, I really prefer the analog EQs regardless of the Q setting. Most of the time they're much more musical and I think a mouse is the worst way to adjust EQ in general.
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Jun 7, 2007 22:58
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No I didn't make it up. Linear phase EQs are linear phase. There really is no other way you need to interpret it. I already said there are digital EQs that attempt to mimic analogue EQ behaviour. Think of them as virtual analogue EQs. I specifically said to ignore those if you are looking for linear phase EQs. As for the difference. Its clearly obvious if you use something like Izotope Ozone and you switch back and forth between its linear and non linear phase paragraphic EQ which you can do with one click. As for the information, alot of it is based on what Izotope recommends you do with Ozone. Its pretty sound advice for general purposes. 'musicality' is a bullshit term when used to describe tools like EQs and compressors. It depends entirely on what application you use them for. Even the most 'musical' compressor for instance can be made to sound very 'unmusical' if you apply enough gain limitation. There are some instances where you don't want a phase shift or at the very least want to keep it minimal. I used to hate LinEQ until I 'got the point of it.' But once again we dive into the 'highend' in a home recording thread. Nobody here is going to have an Avalon. If you do thats great - use whichever tool you feel most comfortable with. Alot of people here are going to be using VST based effects for their post processing - the very idea of using outboard without having stupidly expensive convertors is just silly. All software effects are digital but many are designed to mimic analog filtering using digital algorithyms. Thats why the distinction is sometimes referred to (read: simplified to) analogue and digital EQs. This information is intended to make some sort of sense for people who are confronted with a billion VST EQs and don't know which one would be better suited for a particular application. In the end there are no hard and fast rules and even those can be broken if you know them well enough. But as a starting point, its a good system to get your bearing and I'd like to see you elaborate on how this is demonstrably flawed - its you against Izotope at least on this score. and who says you have to use a mouse to operate a software EQ? Jesus, just assign the relevant CC values to some rotaries on your midi controller and voila - all rotary operation. Besides, if you are designing an EQ to be used on a computer it makes sense to make it functional for use with a mouse and keyboard. Alot of people say that analogue synthesizers are not meant to be programmed with a mouse and keyboard but Native Instruments Absynth proved that idea was bullshit. The problem of course is with GUIs that look and are designed to be programmed exactly like rotary control surfaces yet have to be operated with a mouse. Anyone who has struggled to mouse over a rotary on a synth like pro53 will know that its GUI was designed to look exactly like the hardware - it was never designed to be easy to operate using a mouse. Which of course is a problem considering NI were selling that product to people who will always be operating it on a computer. However, there are many ways its GUI could be altered to make it a cinche for mouse and keyboard operation. Allowing you to input numbers using the numpad for instance and having all variables have a numerical range display. Even without it you can use it exactly like hardware by mapping all the variables to a midi control surface. So what exactly is the problem? WanderingKid fucked around with this message at 11:11 on Jun 8, 2007 |
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Jun 8, 2007 10:45
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I think you bring up some good points but my specific criticism was with this statement:quote:So yeah - digital EQs for precise surgical cuts and boosts. Analogue EQs for broad sweeping cuts/boosts. By your own logic a home studio isn't going to be equipped to use analog EQs as outboard gear, so what exactly are you trying to say here? Don't use digital EQs with wide Q settings? What do you do when you don't have a console and good converters then? I have the full waves bundle so I know what you're talking about with linear phase EQing, I just think you're equating something here that isn't necessarily true. Some analog EQs have a lot of phase smear, others don't. Depending on the application sometimes phase smear sounds good. Now I Haven't used the IzoTope plugins so maybe you'd care to enlighten me as to what you're talking about there. It could be that their reccomendations for using their plugin may not in fact relate to all eqs in general, but I couldn't say with an informed opinion. All I know is that in general I don't see digital EQs as being good for one application and analog EQs being good for another, I see it as there are some good digital eqs and some good analog eqs and in the end whatever sounds the best from what's availabe is what I'll use. also quote:'musicality' is a bullshit term when used to describe tools like EQs and compressors. It depends entirely on what application you use them for. Even the most 'musical' compressor for instance can be made to sound very 'unmusical' if you apply enough gain limitation. I think it's very important to always try and think of your tools in context, and since most of my work is done with music I try to think of how effective my tools are at musically affecting sound. You're right, an empirical labs distressor can be used in non-musical ways but I consider it capable of being very musical. This is of course an opinion and there is no spec to measure it other than a lot of people agree with me. But if we try to divorce ourselves from our subjectivity when working with music I think we do ourselves a disservice. Now as for mapping midi controllers, I'm going to call you out on this one. Midi controllers have a range of 0-127, would you like to explain to me how you're going to get a clean sweep of frequencies by dividing a 20khz range into 127 values? I'm sorry but even the digidesign controllers (I own a control 24 btw) don't give you adequate control for dialing in an EQ, and they have 16,129 values. EQ is where analog has a much greater advantage and I think even the most scientific minds can agree that right now hardware controllers for digital equipment don't let you both broadly and minutely tune in the way an analog knob does. When using our amek at my studio I can with a single knob run the entire frequency spectrum and at the same time barely nudge and squeeze until I'm dead on. With an EQ plugin at best I can activate a "minute" adjustement feature but I've never had that form of control be as intuitive as a knob.
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Jun 8, 2007 17:44
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RivensBitch posted:I think you bring up some good points but my specific criticism was with this statement: You may notice that I prefixed pretty much everything in my post with 'I tend to...' because its a habit I've gotten into that seems to be logical and works for me. It might work for you. Try it out. For those without any sort of system I believe its a nice set of simple ideas to start off the subject before you go off and experiment. You are of course correct in saying that analogue EQs vary in the degree of phase smear. And that there are nice sounding analogue EQs and not so nice sounding analogue EQs (in my and your opinion). The idea of linear phase EQs being appropriate for certain basic situations is fairly solid. You would use a linear phase filter for when you want to minimise or eliminate phase smear/comb filtering in stereo sounds. You would use a non linear phase filter if you didn't care either way. Some people find that phase smearing sound quite pleasant in certain situations if slightly unpredictable. Usually if I am making broad and shallow boosts on a paragraphic EQ you preserve alot of the harmonic structure of the sound so you don't really change the relationship between harmonic intervals that much (unless you boost using loads of gain). So I find these broad shallow boosts to be good for emphasising a broad characteristic of a sound - more like an emphasis. I find using EQs with extremely non linear phase characteristics can often sound quite nice for situations like that. When I am making very sharp, narrow cuts using alot of gain I often find that there are several things to think about : 1) Notching one channel of a stereo pair can be a pretty big source of comb filtering if you want it. 2) You can eliminate certain harmonics altogether by setting the EQ point to one of the loud 'spikes' in amplitude (harmonic interval or partial) and notching it out completely using a narrow Q cut with alot of -gain. This is the trick I described above your reply. 3) If you do point 2 to harmonic sounds and you do it to the lowest harmonic (the fundamental) then you can change the pitch reference of a sound. The best way I can explain this is with the 303s resonant filter. If you turn the resonance all the way up and sweep the filter's cutoff on a single monotone note played on all 16 steps of the sequencer, the tone will subtely seem to rise and fall in pitch as you open/close the filter. Thats because you apply alot of gain to the frequency around the cutoff point. When you sweep it over a harmonic peak it accentuates it and can change the pitch reference of the entire sound if theres enough resonance. 4) Hard notching out alot of harmonics in a series can make a harmonic sound seem very very unnatural. Its the same with extreme high/low/band passing filters. For various reasons I seem to use this notch EQ trick to diminish harmonics or amplitude spikes which would otherwise eat headroom or cause a clip. Sometimes you can stack several sounds (say a clap and a snare drum) in the mix and bring out certain harmonics in one or the other to change things like the pitch reference of a sound or to eliminate harmonic peaks which sound unpleasant or painful at normal listening level. So I happen to use this trick alot for corrective EQ - where I don't want to cause comb filtering or phase smearing and I just want to nuke an amplitude spike because its annoying me. Linear Phase EQs are great for that. It seems to me that you have used Waves LinEQ and was probably left with the same impression I was - I hated it at first because it sounds DEADPAN. Even moderate degrees of subtractive EQ can totally kill an interesting sound stone dead. Then after a couple of months of using Izotope Ozone's linear phase EQ and comparing it with the analogue modelled EQ I just kind of got the point of it and where I could play to its strengths. If you wanted something that sounds like it exploded out of a vaccuum tube amp that costs 10 grand and sounds a million dollars - you aren't going to get it with LinEQ. But you would be missing the point of it if you dismissed it on this criteria. However, there are no rules and most of them were made to be broken anyway. Its just a useful starting point for people getting under the hood of EQ and want to try something out before they start bending/breaking rules.
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Jun 8, 2007 22:21
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RivensBitch posted:Now as for mapping midi controllers, I'm going to call you out on this one. Midi controllers have a range of 0-127, would you like to explain to me how you're going to get a clean sweep of frequencies by dividing a 20khz range into 127 values? I'm sorry but even the digidesign controllers (I own a control 24 btw) don't give you adequate control for dialing in an EQ, and they have 16,129 values. EQ is where analog has a much greater advantage and I think even the most scientific minds can agree that right now hardware controllers for digital equipment don't let you both broadly and minutely tune in the way an analog knob does. When using our amek at my studio I can with a single knob run the entire frequency spectrum and at the same time barely nudge and squeeze until I'm dead on. With an EQ plugin at best I can activate a "minute" adjustement feature but I've never had that form of control be as intuitive as a knob. Because even though MIDI can only have 128 discrete levels of sensitivity, you can still rapidly interpolate between those values. The more times per second the smoother the result. This for instance is how the Access Virus has a digital filter which doesn't audibly step. In fact, most virtual analogue synthesizers built within the last 10 years has a filter which constantly interpolates between steps for smooth operation. If it weren't for the perculiar behaviour of its resonance I would defy anyone to call out the Virus filter amidst a crowd of analogue polysynth filters. That of course is a totally loaded statement since pretty much all of them have filters with different phase characteristics, different saturation properties etc etc making it pretty much guesswork unless you have precise knowledge of how some of these filters work. Hell, the only virtual analogue synth I have ever used which had a zipper filter was the Roland SH-32. I hear the Sequential Prophet 600 has a zipper filter too but I can't comment on that one - I thought it was analogue. Have you never noticed before that you can divide up 19,980hz (the entire sweep range of a software paragraphic EQ like Voxengo GlissEQ) into 128 bands which equates to each band being roughly 130hz in width. Have you not noticed that you can sweep the cutoff and it wont actually step in 130hz intervals? Either from mouse operation or when bound to a MIDI controller? Now on a software EQ you can overlay spectrums and zoom in to something ridiculous. And on some of them you can input frequency and amplitude values to several decimal places. I think that Native Instruments Absynth allows you to input filter cutoff and resonance settings to 3 decimal places? Using a computer savant's best friend - the numpad. Its not infinitely continuous like analogue is but thats a lorra discrete intervals. When you have this much resolution I honestly can't tell the difference. How many different discrete values can you get out of 64 bits? millions? billions? Will you ever use those numbers or be able to tell the difference between 200.001hz and 200.002hz? No but thats the point. - I know what you mean about musicality in essence because I've used a Distressor before. It sounds absolutely class and its a scream to use. But I can't describe what it does much better than saying I enjoyed using it and if I had the money I'd have one myself. Beyond that you make up your own mind about what that might mean. WanderingKid fucked around with this message at 23:00 on Jun 8, 2007 |
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Jun 8, 2007 22:23
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I understand the mechanics behind trying to work around digital controls for digital plugins, I think my point was that it's still trying to recreate what comes naturally with an analog knob, and the feel to me is very important in the same way that the fretboard on a guitar is important to a guitarist. The results I get when I EQ that way are just better, and I credit both the quality but also the control.
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Jun 9, 2007 00:41
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WanderingKid posted:So I happen to use this trick alot for corrective EQ - where I don't want to cause comb filtering or phase smearing and I just want to nuke an amplitude spike because its annoying me. Linear Phase EQs are great for that. Staying out of this discussion but seconding this one "No! Bad harmonic! Be gone!"
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Jun 9, 2007 00:48
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I've never used a linear phase EQ. From looking around it seems to me like in any application where you don't have full-on delay compensation (and any application where a 30ms delay is unacceptable) linear phase EQ is more trouble than it's worth. It takes so long it could easily put the entire track out of phase, instead of just slightly smearing the contents of the track like a traditional EQ would do. This talk of harmonics brings up an interesting thing I've learned about EQ though: often when you try to do something like soften the crack of a snare or get an annoying honk out of a voice, it helps to adjust up and down an octave from where you think the problem is. So if you've swept around and decided that 400Hz is annoying you in a vocal track, cut 400 a bit and then cut 200Hz and 800Hz maybe half as much and see if that cleans things up a bit too. This works almost magically sometimes when I'm ringing out monitors and 1.6KHz seems to be ready to feedback but when I pull more out it sounds funny. If I pull a little bit of 3.15KHz and/or 800Hz and suddenly everything is OK. Sometimes it doesn't do anything especially good though, so trust your ears.
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Jun 9, 2007 01:34
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I'm recently making a purchase on a Macbook Pro, and I'm trying to decide what software to get. I've had experience with Cubase, and I like it a lot, but then there's also Logic, which appeals to me, but I've never used. Could anyone with experience with both these recommend one over the other?
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Jun 9, 2007 20:10
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logic is MUCH more expensive
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Jun 9, 2007 20:45
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RivensBitch posted:logic is MUCH more expensive Logic Express is 100 bucks. If you're resourceful (go to a college bookstore that sells Apple software) you can get Pro for $300-$350. I use Logic Pro and like it a lot. A friend had Cubase SX and I used that quite a bit and didn't like it a whole lot. It had a few plugins I liked, and I like how the dynamics plugin is integrated, but aside from that Logic's plugins sound better and are easier to use. Logic's interface is hard to wrap your head around at first... you'll probably have to spend some time making templates for yourself based on how you like to work and what you're used to. Once you've got that down, you should like it.
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Jun 10, 2007 08:38
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starbucks972 posted:Could anyone with experience with both these recommend one over the other? Ableton Live out of loving nowhere  Used Cubase, eventually came to hate it because just you try assigning a MIDI controller to a softsynth which doesn't have MIDI learn - you're quite simply hosed. The one concept which could've made it all really great doesn't have much support from Steiny itself (user panels). Had a brief glance at Logic which did some things a lot easier and I was very impressed with the plugins it came with. edit: re: my question above: I'm going with the Onyx because it's also a mixer at the end of the day 
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Jun 10, 2007 20:08
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I'm thinking of making a small recording setup in my house to record my band/other local bands. The setup I have in mind is an Alesis Multimix 8usb with mics/moniters/all that stuff. I will be using Adobe Audition 2.0. Are they compatible? Is this good for starting out? I've heard the Alesis Multimix 8 is good, is this true? Thanks
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Jun 11, 2007 05:10
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wixard posted:I've never used a linear phase EQ. From looking around it seems to me like in any application where you don't have full-on delay compensation (and any application where a 30ms delay is unacceptable) linear phase EQ is more trouble than it's worth. It takes so long it could easily put the entire track out of phase, instead of just slightly smearing the contents of the track like a traditional EQ would do. Aye. The thing about alot of drums is that they are atonal. Most membranophones dont vibrate in the manner a string would for example and instead of creating an ordered series of harmonics you get a quite complex series of partials. Sound on Sound did an awesome guide which should be available on their site called 'Synth Secrets' - The author explores the nature of membranophonic instruments and attempts to recreate the partials of a Timpani (I think) using a DX7. He does a good job too. I have nowhere near that level of knowledge on the subject but I figure its worth learning since you can start developing EQ routines and shortcuts that work consistantly for atonal instruments too. Some of the engineers in this field are truly insane like that - they have this mathematical understanding of sound design which makes me look like a total amateur. With regards to the latency - its not exclusively related to linear phase EQs. Convolution reverbs tend to have alot of latency too. Thats just part and parcel of working with software to a CPU limit. There are ways of working around this. For example I have separate instruments in separate project files and mix using several instances of FL Studio open. And I have to bounce alot and chop off the dead silence at the start of each wav. For alot of people this is a pain in the arse and for people like you that have regular access to full on pro tools rigs theres absolutely no point or imperative in learning to deal with this kind of problem when you don't have to with pro tools (0 latency and great sounding plugins wahoo!). If you don't have a protools rig or an awesome studio with awesome converters and outboard then you gotta bounce alot. Sure its a pain but look on the bright side, it costs alot less. Think of it as the cost of convenience.
WanderingKid fucked around with this message at 13:31 on Jun 11, 2007 |
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Jun 11, 2007 12:57
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