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Oi, don't be dissin' the fruit. It fuckin' owns.  Edit: Eh, been reading the TC Support forums and it looks like alot of people are having the same issues with Fabrik-C/R and not just XP SP-3 users either. There was 1 guy on a mac and about 10 people using the Powercore version. WanderingKid fucked around with this message at 01:20 on Jun 13, 2010 |
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Jun 11, 2010 16:51
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Can you use Core Audio driver in a Boot Camp Windows 7 machine? If so I guess I'll sell my Xpander when I get it fixed up and buy an MPB.
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Jun 13, 2010 11:13
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Octopre is actually really decent. Focusrite get so much flack for their platinum stuff but I think alot of it is mud that seems to have stuck to Focusrite all the way up to the blue series stuff which the gaggle say is 'boring'. The reds are 'overpriced and boring'. The greens are 'unreliable, overpriced, overrated and boring.' What a load of tosh. I think most of it is just reiterating what someone said before them and god knows where it all started... :< Edit: By the way, I nearly poo poo myself when I found out Focusrite were selling Isa Ones for less than 500 bucks. Thats a killer preamp for that kind of money. WanderingKid fucked around with this message at 20:49 on Jun 16, 2010 |
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Jun 16, 2010 20:46
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poo poo, if you want to go down that road then AMS Neve isn't Neve either. And Fender isn't Fender when the man himself sold his name in the 60s. Hell G&L isn't even Fender now that hes been dead for 20 years. Part of buying the name of something is buying the client base that comes with it. The new owners paid for the focusrite name, the Neve association and I'll say straight up again that the platinum focusrite pres are decent for the money. We are talking a completely different target buyer here than the guys in the market for 110s and 115s so of course they would be disappointed. Yeah the IC thing is a bit misleading but those ICs are in all the new focusrite stuff and it doesn't necessarily make a bad product even if its easier to believe that to be the case. Most of the hysterics come from crazies who for some reason are annoyed that focusrite is no longer a boutique high end British electronics shop and actually sells products that normal people can afford. That opinion seems to have rubbed off on their entire current product line including the good stuff. One thing I will agree on is that Focusrite is very unfashionable amongst the king nerds. WanderingKid fucked around with this message at 00:34 on Jun 17, 2010 |
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Jun 17, 2010 00:01
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RivensBitch posted:Here's the AES paper explaining what "JET" is: Errr thats just a phase locked loop circuit. As in, the same whats in pretty much every soundcard?
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Jun 18, 2010 05:56
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I was skimming for sure because its very long but yes. I paid close attention to the section called '4. The New Technology' and whilst theres alot of jargon, I can't really see how JET anti-jitter is anything other than a dual stage PLL circuit + technobabble? What that paper does illustrate to me is that jitter is fundamentally an engineering problem for the people that engineer converters. Jitter has become this buzzword in the last few years amongst home recordists and the like but I don't think consumers like us really should think of or worry about poo poo like this. Even the worst converters produced in the past 10 years are in engineering terms miles ahead of anything available to the super pros of the late 80s. Its the same with folks who worry about cable capacitance and dither. Those are problems for gear designers not musicians. WanderingKid fucked around with this message at 11:00 on Jun 18, 2010 |
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Jun 18, 2010 10:57
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I used to mix only with software and its easy to run softsynths right up to the red line on a full scale dB meter. But as soon as I started getting hardware synths I could max out the volume on the synth and run the output into a balanced soundcard line in and I would get in the ballpark of -20 to -10dBFS. My Xpander is broken but I plugged it in yesterday and its pushing -12dB in FL Studio with the volume controls on the synth totally maxed out. I guess you can run it into the neutrik inputs on the front of the Konnekt 48 with the preamps but I never tried it. My Virus sounds shocking when you try to full scale red line it even though its designed to work at +4dBu. You can get loads of volume out of it by using feedback1 or 2 reverb and mixing in tonnes of wet signal but it just starts breaking up like its clipping bad even though its miles under the 0dB line in FL Studio's meters. FL Studio meters really suck though.
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Jun 18, 2010 13:21
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Windows XP sucks donkey balls though compared to Vista and 7. :|
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Jun 24, 2010 13:56
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Couldn't you just run Windows 7 in bootcamp to take it for a test drive? Most of the stuff I use works fine with Windows 7 and Vista. At least, I haven't hit any walls re: compatibility or anything like that. Plus I'm used to stuff like UAC now and honestly, you have to be insane to be running a PC in this day and age without it. They worked it into Windows 7 alot more elegantly than in Vista.
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Jun 24, 2010 19:44
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Nahh I'm talking about user access control. WinXP is like swiss cheese when it comes to security.
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Jun 24, 2010 20:23
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Thats screwball logic though. It looks like it makes sense but its really just nuts. The kind of noise level we are talking about in digital systems is nothing compared to your analogue gear. I mean all of my analogue gear is noisy. Crank the main outs on my Xpander and you can hear it. Even my JD-990 (a digital synth) is noisy as hell, especially patches with the phase shifter in it which produces this constant swishing noise even when theres no signal. I sold my Juno-6 a while back but if you put the chorus on that thing you can hear the 'shhhhhhh' sound. So what if you record it 12dB hotter? You just get 12dB more Juno chorus hiss, 12dB more JD-990 phaser noise and 12dB more oddball chirping Xpander weirdness. All of which is about 10 times more significant than digital system noise anyway. And after all that you are still going to trim all your channel inputs because you got no headroom. It just seems like a whole tonne of effort for nothing.
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Jul 4, 2010 21:58
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Mics? Hell thats even worse. You aren't going to eliminate ambient noise completely, even in a super quiet room and then theres noise generated by the electronics in the preamp etc all of which is way more significant than the digital noise floor. I agree that good gainstaging and elimination of common electrical problems can minimise the noise from analogue gear. But they are all designed to work best around 1.23 Volts RMS anyway and when I stick that into the line ins on my soundcard theres still *loads* off the top of the fader before I'm touching 0dB full scale. Your compressor example I don't get. 20 to 30dB off of peak signal level is 20 to 30dB off of peak signal level. What does it matter if you come in 12dB hotter at your soundcard inputs when the peak signal reference is the same? The 4 least significant bits of a 24 bit conversion are realistically going to be noise anyway. I get where you are coming from and it looks like it makes sense but its actually insane. Its just like the idea that digital synths need to have ever more bits and higher sampling rates than what we have to get closer to analogue sound. It seems like it makes sense because how do you make something that is discrete value/discrete time into something thats infinitely continuous? You keep on adding more and more discrete resolution until the difference isn't noticeable anymore right? But we are already at the stage where most, if not all of the noise byproducts of quantization are either inaudible or cleverly shifted into an inaudible range and all you are doing by having 128 bits and DXD+ sampling rates is increasing sampling inaccuracy and saddling yourself with a massive computational burden. WanderingKid fucked around with this message at 09:17 on Jul 5, 2010 |
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Jul 5, 2010 09:00
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RivensBitch posted:Either way, if you still have appreciable noise introduced by your analog gear when tracking things like drums of vocals, you've got a problem. If you don't, then what are we arguing about? I'm just saying that the whole advantage of going digital is that noise inherent in digital systems is insignificant compared to the kind of noise you get with analogue. So the idea of running a signal into a soundcard thats so hot its about to clip is kind of pointless and its not going to get you any more appreciable signal relative to noise floor. Not when the noise floor of most prosumer ADCs is still less than -100dB. Basically everything you run into an analogue input on your soundcard is going to have more noise. And its even more pointless because you need to trim your levels anyway so have some headroom to mix with. WanderingKid fucked around with this message at 20:42 on Jul 5, 2010 |
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Jul 5, 2010 20:40
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Pretty much everything on the issue that needs to be said already has been here. That forum is mostly full of headcases but that thread gets the point across fairly well and many of the people doing the talking are not wing nuts (for a change). For some more on noise in digital systems check out some of Dan Lavry's posts on the issue. He also posted a bunch of important stuff about AD noise floors here. Heres some more from prosoundweb with some contributions from John Hodgson (the guy who wrote the G-Force plugins). I'm not an expert on the subject since I'm just a dude who makes music. Its clear I'm not explaining the concepts very well so its probably better if you hear it straight from the horse's mouth. WanderingKid fucked around with this message at 13:39 on Jul 6, 2010 |
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Jul 6, 2010 13:31
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Rivensbitch et all aren't doing anything that is inherently wrong. Theres nothing wrong with recording as hot as they can and trimming the level before it hits a soundcard input or desk. The guy in the first link mentioned how he works pro tools sessions all the time where all the channels are recorded hot as hell and when he wants to spread them out on an SSL, he has to trim all the inputs because all his VU meters are pegged. You can do that if you have to. But you aren't lowering SNR by any degree that matters by simply recording -12 to -20dB below full scale to begin with and you don't have to trim later. And all your hardware sounds better because it was designed to operate around 0 VU (volume units) anyway. Above 0 VU you have quite alot of headroom before the signal turns to poo poo. In digital systems, the signal turns to poo poo precisely at 0dB full scale. So what most people are doing nowadays is calibrating their meters and gain structuring so that they have some headroom in their DAW before they clip. It doesn't matter whether you do it pre recording or post recording. It only matters that you do it otherwise how the hell are you gonna mix with no headroom? Thats not the point of contention however and I'm pretty sure we all agree on these points. The point of contention arises from this idea that recording super hot gives you higher signal to noise ratio and thats not true because the noise introduced from the digital part of the system - the converters is insignificant next to noise generated from the electronics in whatever box you are recording. If you are getting audible 60hz hum and you record quiet and then later on try to squeeze +20dB out of it in your DAW then yeah of course you are going to get +20dB more 60hz hum. Thats not a case for recording stupidly hot to begin with. Thats a case for trying to kill a ground loop which is causing an inordinate amount of 60hz hum. If you pump volume like a motherfucker then you will notice more noisy byproducts of your gear. Hell if you pump volume like a motherfucker it is possible to hear quantization noise in a 16 bit recording under very specific conditions and that level of noise is almost insignificant to begin with. Follow the izotope dither guide if you want to check it out yourself but it quickly becomes apparent that nobody actually listens to music like that and you won't be able to resolve quantisation noise from more significant sources of broadband noise or overwhelming signal. WanderingKid fucked around with this message at 16:21 on Jul 6, 2010 |
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Jul 6, 2010 16:09
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I used to do that too. When faced with DAW meters though your natural inclination is to go with green = go, yellow = brace yourself and red = stop. So noobs naturally find themselves mixing to the red line and squashing the poo poo out of everything with compressors because they got no headroom. You gotta create it somewhere I suppose. Shame it tooks years before us bedroom warriors got the message from some guy who works an SSL all day long. You mean, you have ~20dB headroom over the STOP sign on your meters? Where you been all my life baby?
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Jul 6, 2010 17:22
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RivensBitch posted:I'm going to actually find some time to record some test tones at different levels and then normalize them and flip 180 degrees to really get down to what kind of noise might be introduced at the ADC. If WanderingKid and Hog are right, there shouldn't be any appreciable noise difference when recording at 0dB vs -20dB. Most of that has already been worked out for you. TC publish the noise figures for the ADC and DAC stage if you want to go have a look. Emu claims their 0404 has 111dB SNR on the ADC side. That pretty insane. Its just impossible for any analog signal generator or signal processor to produce less noise so its unlikely that the converter stage is going to be the weak link in the chain so to speak unless you digitally create all your sounds and mix it all in a computer. Your preamps will put out alot more noise than your ADC. Most soundcards these days have pretty ridiculous noise figures. Yoozer posted:http://www.gearslutz.com/board/so-m...s-restored.html Beat you. :O
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Jul 6, 2010 22:20
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Isn't this just testing how noisy and non linear preamps are at different amounts of gain? I mean I'm just a dude with a couple of synthesizers and a guitar. I don't even have instruments sensitive enough to measure the noise of converter and preamp stages nor do I have the electronics know how to do it properly. The only spectrum analyser I have is SPAN and that only goes down to -72dB. I just know what engineers teach me for free in their own time and one of the fundamental basics is that digital systems are extremely intolerant of noise and interference. Much more so than analog systems. I mean if you can explain how this is supposed to work and what its attempting to prove I'll help any which way I can but yeah. I'm just a guy with a guitar. Thats why I just linked to posts by Jon Hodgson and Dan Lavry because evidently I'm not paraphrasing them very well.
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Jul 7, 2010 11:04
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I'm gonna try to sum up my position in way that makes sense from a musician's perspective. when you work on a nice desk with a bunch of VU meters theres a red line at 0 VU and if you tick that its not the end of the world because depending on the desk, you got maybe up to +25dB above that before your signal turns to dogshit. In your DAW you tick the red line and you are fine but if you go over it you will clip which is the point at which your signal instantly turns to dogshit. I understand why DAWs have full scale peaking meters instead of the old style VU averaging meters because in digital audio you can't use an averaging meter where there is no overload margin. Its either overloading (clipping) or its not. If you use averaging meters to mix, your meters won't tell you when you clip. So on a nice analog console you got a good deal of headroom (maybe about 20dB) above 0 VU to mix with. In a DAW mixer you can have tonnes of headroom (potentially alot more than 20dB) but its under 0dB full scale. I guess what people are doing now with digital mixing is to make their DAW mixer a bit like a kick rear end desk. You take 20dB off the top of your peaking meters and call -20dB full scale the equivalent of 0 VU. Theres obvious differences between averaging and peaking so its not the same and it wont get the same results but you are worried more about clips in digital audio so it makes sense. The second bit is about gain structure. So if you distort the ever living gently caress out of your guitar and then record it into your soundcard, its going to sound distorted no matter how much you pull down the channel gain. Thats obvious right? If you get your mic and drive your preamp super hard so its hissing all over the place, thats gonna show in the recording. so if you want a super hot, distorted sound thats fine and you know what to do. You drive your amp into its overload margin (saturation) or beyond it and you record it that way. When you mix it in Cubase or whatever you can trim your channel inputs so you have your 20dB of headroom and you can mix it easy. Thats fine. But why would you record everything as hot as it can possibly go? The reason most people use is that you get more signal to noise ratio or that you need to use all your 24 'bits' which is an idea that I don't buy, based on the information I'm getting from Paul Frindle and others. WanderingKid fucked around with this message at 12:03 on Jul 7, 2010 |
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Jul 7, 2010 11:45
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I'm not asking you to accept anything as gospel. Read it or don't. Take what you can away from it. Or don't. I've already said I'd listen to the wavs when I get home because I don't have speakers on my work computer. Reserving judgement on the test until I can hear it.
WanderingKid fucked around with this message at 16:56 on Jul 7, 2010 |
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Jul 7, 2010 16:29
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I had to go bag the Wavelab 5 demo and since it was a quiet day at work, I toyed around with it at the office. No speakers again but all I'm seeing is noise floor that scales with peak signal. You have about 12 bits of useful signal and the rest is noise on both mono wavs. The summed wav is stereo for some reason which I don't know what to make of yet. I suspect the summed file is botched because the residue impulse gets higher in amplitude the further into the recording which may indicate that you didn't line the two mono wavs up to the exact sample? I still don't understand what this test is meant to prove. That if you add +17.5dB of gain you get (surprise) +17.5dB of gain?
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Jul 9, 2010 17:08
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But there is no advantage in recording as hot as possible so thats a loaded question. The idea of recording super hot comes from the 16 bit days and is a defunct habit that some people still do even though there is no logical basis for it. Explanations for this are given by Paul Frindle and Skip Burrows here and here. I've quoted the most relevant passages below. (Skip's spelling is rather atrocious and is copy pasted as is.)Skip Burrows posted:
Paul Frindle posted:Ok one final word about converters... Theres a little bit more with Bob Katz chiming in here. I admit reading through the entire thread is a bit much so I'll just highlight the most relevant posts addressing this issue. But back to your example. I can't get them to null but they are very very slightly different lengths. The first impulse is 1 sample longer than the other and I have no idea why. Getting these files to null is completely tangential to what we are talking about in either case. WanderingKid fucked around with this message at 20:29 on Jul 9, 2010 |
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Jul 9, 2010 20:25
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I feel like I'm beating a dead horse but all of this information is out there and has been for years. You can literally google any of the names I mentioned + bits + dynamic range + noise floor and you will get a tonne of useful information which (of course) is of engineering concern. Heres another one. Whilst its helpful for people with an interest in the science of sound and digital audio it is nonetheless almost completely irrelevant to average joe musician.
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Jul 9, 2010 20:38
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How many channels do you need? Preamps? Does it have to be USB? etc. etc. You can probably get a second hand EMU 1820M or 1616M for less than 300 bucks and those things are amazing. 1820M is firewire and 1616M is PCMCIA though.
WanderingKid fucked around with this message at 04:04 on Jul 10, 2010 |
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Jul 10, 2010 04:02
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Guess you should look at Echo Audiofire 4 then. Its firewire (6 pin bus powered) though. I guess I'm looking in the direction of Echo because they got proper 64 bit drivers, they have a pretty good rep for never having any problems and in most polls, Echo is miles ahead, at the top with RME in terms of most users that are happy with their purchase. So they have to be doing something right eh?
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Jul 10, 2010 04:17
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On the laptop hunt I've hit another dead end. Sooo many lappys have ricoh firewire chipsets with known DPC latency problems and it looks like the only ones that have TI chipsets are the HPs and Macbook Pros. We get screwed on HP screens over here so it looks like I'm just gonna have to bite the bullet and ewwww, buy a Crapple Mac. 
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Jul 28, 2010 13:36
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In all seriousness I'd be just grand with a Macbook Pro. Build quality is excellent. Screen is excellent. Keyboard is excellent even if it has those terrible half sized cursor keys that are oh so fashionable nowadays and no numpad (so working excel is probably a nightmare). The thing that kills it for me is that its just so freakishly expensive. You can CTO a Dell or a Sony with better specs for half the money. You'll probably be in DCP hell unless you disable everything but think of what you can do with the cash left over. If necessary you can spend it on therapy or anger management or whatever else it takes to get through the day without taking a fat one up the bum from Steve Jobs.
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Jul 28, 2010 18:50
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Thirst for Savings posted:M-Audio seems to be hit or miss with their product. I was just curious if they would be better for monitoring than my Swan M10s. Which I love, but I should have made a more conscious purchase for monitoring. Unless you have a big, treated, very uncluttered room then it doesn't really matter what monitors you use because they will all sound like poo poo and you will get that weird thing where you move your head around and the sound just 'disappears' periodically as you move in and out of null points. When I sold/traded off half my hardware I got to see a few people's home rigs and size them up. All of those guys had been relegated like me to box rooms by their wives, partners, families etc. The monitors were Behringer Truths, some weird silver Tannoys and Adam S2As. I have Dynaudio BM5as in mine. They all just sounded like different variations of poo poo. My Dyns sounded awesome in the listening room where I bought them but it was a big room with controlled acoustics. When I got them home they were mostly unusable for anything critical so I do critical mixing on headphones (Shure SE420s to be exact).
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Aug 1, 2010 01:13
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I do use my Dyns though. Just not for critical listening. I also use a pair of Shure SE310s and Sennheiser CX300s, again not for critical listening. If I move my head a couple inches to the left or right, bass completely disappears when using any speakers because I don't have controlled acoustics in my room. Furthermore its rented accomodation so no permanent alterations can be made to the room or the wrath of my landlord will cometh. I am aware of the problems with headphones in the sense that stereo is handled very oddly since a hard panned sound is only heard in one ear and not the other. Whereas with speakers a hard panned sound is heard in both ears but the distance to one ear is shorter and thus the source is perceived as sounding from a different direction. I'm aware of lack of room ambience too. The point is that these deficiences are consistent and predictable in relation to reference tracks which I frequently use. As long as there is consistency I can sense a pattern and learn to work around it. Just like people learn to mix around that 1500hz megabump on NS10s because its always there. I can't work around bass that continually disappears and reappears when I shift from one rear end cheek to the other or I get up, go for a piss, sit back down a couple inches away from where I was before and my mix sounds different. Headphones for me is the lesser and cheaper of two evils. One thing I wouldn't do though is mix with headphones if you live/work in a noisy area. I'm lucky in that I live out in the country where its deadpan quiet and theres no traffic noise. This way I can monitor in headphones at fairly low volume and I can still get the little details. But as soon as you try to compete with traffic noise, noisy computers, noisy neighbors then you find yourself trying to overpower it all by blasting your ear canal phones at 100+dB SPL and thats how you go deaf. RivensBitch posted:Even if you can't treat your mix room, you can still get a decent reference point with an equilateral near field setup. Is it perfect? Absolutely not. Would a treated room be better? Of course. But I mix in my living room all the time and I would never use my shure E5s for mixing unless I had to keep quiet, and even then I'd always come back and make adjustments with the monitors. Thats nice that you can mix in your living room but thats just not possible for me (and most people I would imagine). It would drive anyone that has to live with you up the walls. My living room is the biggest and nicest sounding room in my house though. I love the way my guitar sounds in my living room anyway. WanderingKid fucked around with this message at 09:57 on Aug 1, 2010 |
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Aug 1, 2010 09:32
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Its a pretty small room. I'd guess 4 metres by 4 metres at the largest with less physical space due to the desk, closet and one corner is a diagonal.
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Aug 1, 2010 20:33
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I dont find bass or sub particularly difficult because its consistent and you can get a sense of proportion by comparing it to a professional track you like the sound of, in the same headphones. The thing thats most weird is visualising a 3 dimensional space because you get an exagerated sense of direction when a sound is coming at you off axis. At the extremes you only hear it in one ear which is quite an odd sensation but again its predictable and you can compare to professional tracks so it is possible to work around it. Its not ideal and if I could use monitors for critical listening in a properly treated room I would but thats just not an option for me as it is with many people. I don't see it as being fundamentally different to people who mix on terrible speakers. Once you become acustomed to where the speaker is hyped you can mix around it as long as the places where its hyped stay constant. After a while you instinctively know that your speakers are deficient between x hz and y khz and hyped between a hz and b hz so you begin to compensate accordingly. Do it often enough and it becomes second nature as long as the goalposts don't change so to speak. I don't know what else to say other than I find shifting sound from my monitors in my room very disconcerting and that I have experienced the same phenomenon to greater or lesser degrees in every small room home studio I've been in, regardless of what monitors were used.
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Aug 1, 2010 23:17
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Yeah I'm thinking of DIYing some acoustic treatment but never got around to it. I've been making do with headphones for years so at this point I'm used to it.
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Aug 2, 2010 00:02
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