MrJoshua

As a matter of interest, what sample rate do you track with? I'll take a guess that it's higher than 44.1 anyway I track at 44.1kHz/24-bit. Through Apogee, Focusrite, and Digidesign converters. I've tried 48 and 96kHz and did several songs that way, and just couldn't tell any difference (audible difference, I mean - the computer was certainly getting loaded down a lot faster!) so I went back to 44.1.

The results would sound better. You can't argue with the fact that mixing at 96 will produce better results, even if only for the fact that you'll run into far less 'graininess' problems when adding DSP at a higher sample rate. And that is fully supported by the laws of physics OK, I keep telling myself that I'm going to stop posting in this thread, and after this post I really really am done. We absolutely can argue with your claim that mixing at 96kHz will produce better results, because it's flat-out not true. It just isn't. And it is NOT supported by any laws of physics. You do NOT get a better representation of a 20kHz signal at 96kHz sampling as opposed to 44.1kHz. You just DON'T. Sampling it at 44.1kHz with a properly-designed converter will reproduce that signal with 100% accuracy, and it just doesn't get any better than that because it CAN'T. Once it's accurate, it's accurate. I know you heard more space around it, a blacker sonic background, more depth, a wide stereo field, a veil came off the speakers, and angels sang in the background but the simple fact of the matter is, there are LOTS of things that cause that reaction in people and a great many of them have been shown to be complete and utter bunk. We're only human, man, and each and every one of us (including myself) is perfectly capable of hearing what we expect to hear. Believe what you want to believe, but I implore you to do some more reading up on converter design and sampling theory. I think you'll be surprised what you might find out, and it could go a long way toward helping you avoid pseudo-scientific sales pitches in the future. Now that I've said that, let me add that if nothing else, I'm glad you've brought this topic up as it's always an interesting one to discuss, at least up to a point. If nobody ever disagreed about anything, none of us would ever learn anything and the world would be awfully boring. So I hope you'll take this for friendly disagreement, which is what it's meant to be, and that you'll keep sharing your opinions on the board. Plenty of room here for varying points of view, and I look forward to debating other topics with you in the future. I just think that we've carried this one about as far as it can go without bringing calculus into it, lol.

That was quite nasty, sorry guys. Do google that book tho, I know I'm not talking garbage and I can guarantee you that if you do an AB recording session using a 48kHz and a 96kHz system, the 96kHz system recording will sound better, even bounced down to disk @16 bit 44.1Khz. The difference may be small, but then why bother developing 192kHz systems if it makes no difference? It is analogous to the difference between 1" and 2" tape. It isn't analogous to 1" and 2" tape at all. It isn't even similar. As to why it was developed, well, you'll have to ask the guys in the marketing department. I've done numerous A/B checks recording at rates from 44.1kHz to 96kHz, and if you have a decent converter it doesn't make a single bit of difference. Not even a small one. edit: You keep saying that time is a factor. Of course it's a factor. Why do you think we keep talking about frequency?

Wow. Bamboozling ourselves with mathematics? Now who's insulting someone's intelligence? You're using just enough science to back up what you're saying without applying enough of it to realize that you're drawing an erroneous conclusion. Believe what you want to believe. But converters and signal manipulation are nothing BUT mathematics. Until you accept that, you're going to have a hard time getting anywhere.

BTW, I did Nyquist in school too and all I learned from it is that if you're gonna make an AD convertor, you'd better make sure to stick a LPF in there somewhere to kill all freqs above half the sample rate unless you want everything to sound like internet streaming audio circa 1996. How does a 20kHz waveform not become square if you're sampling at 40kHz? Do the little elves make it back into the same shape it was when it went in? Sorry for the deliberate facetiousness but come on guys.... Your sampling rate has to be above twice the frequency, not equal to. I hope I didn't say "equal to" last night when I should have said "above" but I may have - it had been a long day. But if you're sampling a 20kHz waveform and your sampling frequency is above 40kHz ... say, 44.1kHz ... then you're going to get an accurate recreation of the waveform. The facetiousness doesn't change the mathematics.

mozart, grandma, and math have nothing in common. ...other than the letters M and A. Sorry. It's been that kind of day.

It can't possibly do so. And ok maybe it might, I'm not a boffin, but how about 10 samples per 9kHz waveform than 5, that's gotta be a better representation of what went in, no? Nope. There's a whole field of engineering and mathematics behind this, and I've only taken a couple of classes (and college was years ago so I've forgotten quite a bit of it), but once you've reached that 2-samples-per-cycle number, you get an accurate waveform. If you're interested, read up a bit on the Nyquist sampling theorem. Or Nyquist-Shannon, I believe it's called now. Wikipedia's Entry What it basically boils down to is, an analog signal (such as a soundwave, or more accurately, the electrical signal caused by that soundwave hitting a microphone diaphragm) can be represented as a sum of sinusoidal components. Any sinusoidal component can be accurately sampled as long as the sample rate is above twice the frequency of the sinusoid. Thus, any signal can be accurately reproduced as long as the sampling rate is above twice the highest frequency of the signal. Whew. That's very long-winded, isn't it? There's a fair bit of math behind it, and it took me quite a while to see it too, because it seems counterintuitive - it seems like more samples OUGHT to make for a more accurate waveform. But it doesn't. It just makes for a waveform that's just as accurate as the one we already had.

Yes it will, time is a factor. I'd rather have 4 samples per 20kHz waveform than 2 You might rather, but your software doesn't care. It's going to reproduce the same waveform from the 2 samples as it would from the four.

I'm not talking about frequencies that we can't hear. If you set a LPF to 10kHz and record in 96kHz, you are still getting a more accurate encoding of the waveform over time and it is audible. For making CD's, it probably doesn't make any sort of qualitative difference, but if I had the money, I know how I'd be recording. OK, I'm afraid this just isn't true. You'll capture the exact same waveform in this scenario at 44.1kHz, 96kHz, or even at 20kHz. You need a sampling frequency equal to twice the highest frequency you want to capture. That's it. Anything above that will not improve the accuracy of the waveform in any shape, form, or fashion.

I didn't think I heard anything, it sounded better in 96kHz on playback thru both systems. I'm not telling anyone to go off and waste money on a system that can handle 96kHz when 48 or 44.1kHz works just fine for anything that's going onto CD. 96kHz is still a more accurate map of the waveform whichever way you look at it. Don't insult my intelligence like that again please friend. But that's just what I'm saying - no offense intended, but 96kHz will only give you a "more accurate map of the waveform" if you're trying to capture frequencies above 24kHz, and you shouldn't have any of that bouncing around anyway. At anything in the actual audible range you'll get the same waveform from 44.1kHz as you'll get at 192kHz. EXACTLY the same. I apologize if the first post came across as insulting - it wasn't intended to be. Sometimes text doesn't do a very good job of conveying a tone of voice. Furthermore, if you did start capturing frequencies in the 40kHz range ... what then? Microphones generally list their frequency ranges as being in the 20-20kHz band, and even if you have monitors that will reproduce sounds above 20kHz your audience certainly will not unless you distribute exclusively to rabid audiophiles. The point of higher sampling rates relates purely to the hardware design of the converters, as I said before - it simply makes it easier to design a low-pass filter (that filters out any noise at inaudible high frequencies) to put in front of the converter. That's it. On a good, properly-designed converter, there isn't much point in using anything above 44.1kHz.

96kHz doesn't give you a better recording, unless you're using converters with a fairly poorly-designed low-pass filter on the high end. It just doesn't. Regardless of what anyone thought they heard on any A/B tests, or any other anecdotal evidence to the contrary. The only purpose of going to those higher sampling frequencies it to make it easier to design a low-pass filter for the converter. That's it. It doesn't capture any inaudible frequencies that interact with the audio (and even if it did, the microphones you're using can't capture those frequencies, the monitors you're using can't reproduce those frequencies, and none of the audio systems your music will be played on can get anywhere near those frequencies). You're better off spending your money on some good mics than on a computer system that can handle 96kHz.

I normally use 44.1, but at 24-bit.

A week after you finish tracking, nobody's going to care what kind of converters you used. All they'll care about is, does it sound good? If it does, great. Mission accomplished. If it doesn't, take a look at the weak points in the signal chain and see what you can do to make it sound better. You can record awful sounds with an Apogee, and you can make great music with a Mackie board and a couple of ADATs (maybe not as easily, but it can be done). Now let's everybody take a deep breath and remember that we're here to talk about making music, and that converters are only one small part of the overall signal chain. An important part, sure, but then so is the microphone, and the preamp, and let's not forget the source...

Listening through cheap earbud headphones through my laptop integrated soundcard, I don't think I'd kick either file out of a session for being poorly converted.

I'd be very interested to find out exactly what it is that makes cheap converters sound "cheap." Let's face it - AD/DA conversion isn't exactly some esoteric physics-laden niche that only three people in the world understand. It's a pretty basic and simple process of sampling an input signal and storing that value, over and over again, tens of thousands of times in a second. So what's the snag? If the conversion process itself isn't the problem, is it the analog circuitry around the converter? But that should also be a fairly simple electronic circuit, and hardly rocket science. I own four different levels of conversion, speaking price-wise: 1. SM-PRO Audio eight-channel preamp with onboard AD/DA conversion. Very, very cheap. I think it was right around $100. 2. Digidesign Digi002 Rack. Not exactly cheap, but hardly the most expensive converters in the world. 3. The converter card for the Focusrite ISA-428 preamp pack. Eight channels of AD conversion, right around $650 when I bought it (I think it's around $800 now, so call it $100/channel). I count this as more expensive than the Digi on a per-channel-of-conversion basis because it's strictly AD conversion - no Firewire computer interface, no DA, etc., while the Digi does all of these things. 4. Apogee Rosetta 200. 2 channels of AD/DA conversion. Around $1800 new. Call it $450 per channel of conversion (treating the AD and DA as separate entities). Out of those three converters, the only one that actually sucks is the SM-PRO. And it only sucks because it's noisy - you record-enable two tracks and you can start to hear a static buildup. Three or four tracks and it's a very noticeable hiss in the background. It's like a cassette tape that was recorded with the level too low, so you have to crank it up and listen to the hiss when playing it back. The Apogee sounds great. Really fabulous. But so does the Focusrite, at a much lower cost per channel. It stands up very well alongside the Apogee. I use both of them all the time and if I make a recording without taking any notes and then come back a week later, I can't tell which converter I used on it. They both sound great. The Digi? Well, guess what. It sounds good too. I think I can hear a difference between it and the other two, but I'm more than willing to admit that this could easily be in my head because I'm very aware of how much I spent on each piece of gear. And while I don't use it as much as the other two it does get used a very decent amount of time and I've never had anybody listening to a track say "man, that hi-hat sounds like it was recorded through some crappy converters." I have some nice gear, and I enjoy having it. It helps make it easier for me to record great sounds, and that's the goal. But are they vital? Nah. If all I had was an eight-track recorder and a Mackie board I'd still be recording music, and I like to think I'd still be getting good sounds down on tape. But the nice stuff does make it a little easier, I think. But hey, I've been wrong before.

If you're buying a cab new, this is the best deal out there that I've found. $219 for a Vintage 30, G12H30, G12T75, Greenback, Classic Lead 80, Hellatone 30, Hellatone 60, 60L or G12K100 loaded 1X12 = ridiculous! Same speaker choice but 2X12 is $358. http://avatarspeakers.com/ Avatar makes solid guitar cabs. I have one of their 2x12 semi-open-back models with a Vintage 30 and a G12H30. I run an Orange "Tiny Terror" into it, and it sounds fantastic. It's pretty hard to beat for a setup that cost under a grand.

44.1kHz

Is there really a noticeable difference between channel 1 and channel two in your Porticos and RNP's? That sounds like a quality issue if so. There shouldn't be. I'm pretty sure that was a joke. We don't all have the luxury of using a different preamp on every channel.

I'm generally not a huge fan of transformerless mikes in general... give me the iron, and no one gets hurt. Just as an aside, what is it you generally prefer about transformer mics? I've never really made that much of a distinction between the two - as long as the mic sounds good, I'm happy. Lately my favorite vocal mic in my (very limited) cabinet has been the Microtech-Gefell MT71-S, which is transformerless unless I'm very mistaken, and it sounds freaking fantastic. I usually run it through a Great River MP-500NV preamp or a Brick, or for my voice I like it through an API 512c to get a little more edge on the sound...and all of those preamps are transformer-coupled, I believe. So I wonder if the reason I don't make much distinction as to whether or not the mic has a transformer is because I tend to use preamps with transformers. :poke: If that makes any sense. I know plenty about electronics, but not so much about their application to music. So I tend to ask weird questions as part of the learning process.

I just don't think I know enough about you one way or the other, so for now, I'll just assume that you have the recording engineering chops of a Roy Halee or a Bruce Swedien and leave it at that! I'm pretty sure I have the recording chops of a John Doe, or maybe even a Bob From Accounting.

So, if she weighs the same as a duck, she's made of wood! And therefore? ... ...a witch!

I hope you brought enough to share with the whole class! Incidentally, to point out a difference in two mics, last night I re-tracked the vocal part on a song where the singer wasn't happy with the original sound. We originally used a Rode NTK through an API 512c (Apogee Rosetta 200 converter), and as you can probably imagine, it had a very forward, strident sound in the mix. But it also accentuated the nasal qualities of the singer's voice. So, we re-did the track using the Microtech Gefell MT71-S into a Great River MP-500NV (same Rosetta 200 converter), and it captured a much smoother sound with a silky top end. The singer sounded less nasal, and while I'm sure a large part of that was the performance, the mic/pre combination did help. That doesn't mean the Rode is a bad mic. I've had some singers who loved the way it sounded, and sometimes it's a great choice. But for this particular singer it just turned out to be a bad call on my part (made mostly due to the fact that I had just bought it and wanted to try it out, since this wasn't a paying session). There are a lot of decent, reasonably-priced mics out there. Get a few of them that have different strengths and enjoy the ability to mix-and-match with different sources. It's one of the things I enjoy about studio work - finding that right combination that can take a sound from reasonable to magical.

Well, this got a little off topic, didn't it? But I think it's clear that "good mic" doesn't always equal "expensive mic" and vice-versa. The SM-7b is one of my favorite vocal mics; lately, so is the Microtech Gefell MT71-S (I hope I'm remembering the model number correctly). One of them is about a third of the cost of the other, and I couldn't tell you which is the "better" mic because it totally depends on what I'm recording at the time and what kind of sound I'm going for, etc. The SM-57 is one of my favorite mics for snare drums and guitar cabs, and it costs less than a hundred bucks. You don't have to spend thousands and thousands of dollars to get a good-sounding mic. But if you want the absolute tip-top, the very last 1% of possible quality and you're recording in a great facility with a room designed and treated to give the absolute maximum of sound fidelity with players and singers that are at the peak of their game, then yeah, you might find that those uber-high-dollar microphones give you that last little bit of clarity and precision that you need to go from "really good" to "un-freaking-believable." If you're recording in your spare bedroom with your college bandmates and you're going to be uploading the files to your MySpace page ... probably not so much reason for you to spend thousands of dollars on a single microphone.

...Support from 3rd parties has been limited and, where it exists, frequently compromised or simply unworkable... ...which is the cause of every other problem with the OS, basically. Once the recording applications come along that have been designed for Vista and had time to work out bugs, make enhancements for the OS, etc., then things will be fine. Honestly, this kind of furor comes along every time a new OS comes out. When XP came out, I can't count the number of people screaming about how you'd have to pry their 2000 Pro out of their cold, dead hands. And now they're saying the same thing about XP. Why? Because it's had time to mature, third-party software support caught up, and they eventually upgraded to machines that exceeded the bare minimum performance requirements. Quit spazzing out. It's an operating system.

Vista only sucks for recording because third-party programmers (the people who write the music programs) haven't written their code to support it yet. Give it time. Within a year it will be just fine.