WRGKMC

I have several scopes at home I got when a company I worked for was going out of business. They gave me some they were going to throw out. One was a Techtronics Dual trace that lets you read two sources at the same time. The other is a portable. Both work last time I checked. I think I have a third that needs some work. Problem is finding parts and getting them calibrated. For comparing audio waveforms and seeing responnce differences I just use HarBal. Its latest version will allow you to view audio in super fine detail. You can also put up one wave as a source and another as a reference. If the two superimpose closely, you wont be able to hear a difference between the two as far as frequencies go. Noise and frequency variations are clearly visable long before you can hear a difference.

"It's because their filters aren't designed properly to prevent aliasing when you record at 44.1 or even sometimes 48kHz, so you still wind up with some ultrasonic junk in the samples that turns into noise and funkiness in the signal" I think you explaind it pretty much the way I understood it as well. I think in my case running at 48 seems to minimise these artifacts/noise fairly well. I was running at 88 but like you said the space is the issue. This may very well be the filters used. My previous card was very clean in that respect and I recorded at 44 for many years. Since it wasnt upgradeable to XP I finally got two M-Audio 1010LTs. The specs dont go down to 20HZ, not that I use it that low, but I like to do my own trimming if you know what I mean. I cant complain, 16 channels for $100 is about as good as you can get. Any how thanks for the explanations guys. My electronics education began in 74. They included tubes, and digital was still fairly new. I try to keep up with it all. Luckily My day job is all digital electronics. The company spends big bucks training people to keep them up to date, but there again they only go so far. If its something you dont need to know, and you have the tools to troubleshoot it, then they often black box it and leave a big question mark there. Barely any info can be found on certain manufacturers designs and whats out there is often poorly written or misleading on purpose. Its not all that different than audio stuff though, actually when you get into different forms of data compression and how it can be manipulated it gets pretty complex and interesting. I think Digital Audio is lagging pretty far behind Digital photography at this point and most of the improvements in audio are offshoots of the movie industry. I'm just curious to see how it goes in the next 10 years.

Mr Joshua, You seem to be up on this subject, Can you tell me about aliasing and what the effects are. I read up on it but the laymens term kind of eludes me. I read this page several times and I'm interested in how it affects recording in the real world not the nebulace description of it, http://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem I'm curious about this statement --- The anti-aliasing filter is to restrict the bandwidth of the signal to satisfy the condition for proper sampling. Such a restriction works in theory, but is not precisely satisfiable in reality, because realizable filters will always allow some leakage of high frequencies. However, the leakage energy can be made small enough so that the aliasing effects are negligible. That statement led me here. http://en.wikipedia.org/wiki/Aliasing And here http://en.wikipedia.org/wiki/Anti-aliasing_filter I'm just wondering if this is why higher sampeling may sound better with drums/cymbals in my case where I have bleedover on several mics. Somewhere in there it says alising can be improved with higher rates.

Thanks wwwjd , Its just an accumulation of stuff you pick up over the years dealing with this stuff and how it can be viewed or manipulated. I'm not a big retainer of fine details if theres nothing I can do to manipulate it and get the most from it. With interfaces, besides changing a few settings or using an external clock there nothing you can do if it becomes a bottelneck besides replace it. Identifying where your bottelneck is, is the key though. Seems like when you take care of one the bottelneck moves another part of the chain, usually a more expensive one. This in no way should prevent you from being creative or having fun getting the most from what you got. GUBU, If you can scrape up $50 you can buy M-Audio 1010LT cards on EBay if you dig enough and catch a bid on a good day. I got two of them that way and I'm thinling about two more just for the hell of it. They record up to 24/96. The converters arent top of the line but they do the job as good as anything else out the comperable range. You wont have all the knobs and jacks there, You would need a cheap patch bay for easy plugin access, and may have to use a mixer for preamping soft signals, but at loud volumes you can get enough juice to pin the meters micing with the line ins if you need more that two Low Z inputs. Theres Tons of 4 channels and such too. You can get a couple of then cheap if you have the PCI slots and go that way too. All the M-Audio stuff will work with the same driver and can have their clocks linked with they're SPIDF cables. Theres alot of Echo boards out there too. You would have to check their support site and read the manuals to read what involved with theirs though. Theres alot of it at good prices though so waiting to save up $500 or a grand to get something that records at high bit rates isnt nessasary. realtec makes stock computer boards that are 4 track that record 24/196. With 2 of those bough used for say $15 each you could do fairly well testing the higher bit rates. Like I said in my first post, Make sure you got alot of disk space and high processor speeds to utilize them.

Excuse this long post, I pulled this over from a word document I been throwing some thoughts around. --- There is an item here that some of the experienced mixers/musicians might notice mixing at lower bit rates. This is not meant to convince anyone of any truths or myths here its merely an observation having first worked in analog since the late 60s then digital since 94. Please be patient in my explanation and see if you can follom me on this while I tie it together from a musicians standpoint. I am formally trained in electronics and could go dig on the internet for explanations but I

I'll throw my last two cents in here. The difference I HEAR with a higher sampled rate seems to be with live drums recorded, mainly cymbals and snare. Some other sibulance gets there too but I only get it using full frequency condencers. I dont nessasarily consider those items to generate sine waves in the higher frequencies. Some of it resembles something closer to white noise but the frequencies do reach quite high. To have a good recording you dont have to have those frequencies but they sure sound more realistic to me. I do notice it mostly when I am mastering a mixdown and EQing the complete mix. On a 44 mix there is hardly anything usable over 16K. If I bring up the highest frequencies I hear white noise either from a recorded amplifying device, sound card converters or plugin noise, amp his, whatever. Sounds bad and isnt usable so the completed masters lack a certain amount of air. When the tracks have been sampled at higher rates, and all the normal mixing has been done and the mixdown is being masterd, the overall frequency range the analizer detects is extended above 20K. The high frequencies in the 20K region can be boosted without sounding like noise. The frequencies above 20K arent needed so they can be rolled off. The quality of the 16~20K has less hiss, noise and harshness then it does at lower rates. There is alot of debate peoples opinions here and I think its great. I believe the real answer is using your resources to get a superior sounding final product. For me, that littel extra high frequency has been like a godsend getting live drums to sound better and compete better with the other instruments I normally had to carve up to match the drums. In my case getting the high end of the cymbals in the clouds above the vocal range had been the biggest battel. It was trickey work working with 16/44 on many many thousands of recordings over a 10 year period and trying to get super pro results. Not alot of room for error. For me the job was made a bit easier.

Recording analog with tape running high speed captured more details than a tape running consumer speeds. This allowed for a high fidelity working enviornment that cause minimum losses during mix and mastering processing. This is not much different working in digital. The effects of processing digital samples is different but the goal ultimate goal is the same. With a good front end you can record high quality to both a tape and digital running at consumer speeds. I did it for at least 10 years with very good results. Mixing was a bitch though. The settings had to be spot on especially volumes. Running at 24 has alot more flexability and a much more natureal feel to the DAW programs GUI. As far as running 96, I'm afraid my converters on these M-Audio cards dont provide a high enough quality to make a huge difference. The high end is clearer but with good plugs and mixing skills its not always enough to justify the space on all songs. There have been tracks I wish I had recorded at better rates especially with some one take wonders that required alot of processing due to sound issues of various types. Sometimes you can use an EQ as a knife to cut out some bad stuff. When you have to use it as an axe theres usually not alot left to work with. Of course its best to do it over but you dont always have that option.

What I found is the higher rates hold up better when when processed by multiples plugings . I usually keep my higher rates set through exporting to stereo tracks and completing my mastering. Then I will convert it down to 16/44 before burning a CD. In this process the losses due to processing are minimised and the rates allow the signal to retains its clarity and quality. If I were to record at 16/44 and exported that directly to CD the quality would be fine. Its when you pass it through several effects like multiband, limiters, comps, EQs etc and process several times that grainyness begins. The way you need to look at it is how well the signal is preserved during processing, Not how it sounds raw immediately after tracking. Raw tracks are going to pretty much sound the same. After processing the toll taken on a low bit rate track really starts showning its deteriorization in comparison to a higher bit rate file. If you record with a good front end, good mics etc with littel need for processing you're fine at lower rates. Otherwise a higher rate may help some.

You just have to try it and see if the difference is worth the hard drive size. I noticed a crisper sounding live drum set but setteled for 48 for most stuff to save drive space.

The mega files were the issue with me as well. There wasnt enough difference in the sound quality to justify the space. I setteled with 24/48. the 48 was a step above 44 which allows for a littel degration due to processing with plugins. The files arent that much bigger than at 44. Nowheres near as big as 88 or 96

Quote: Originally Posted by WRGKMC The tonal quailites of a mellow acoustic guitar are a poor sources for an test as well. Niether frequency extreme is being challanged. "Agreed, and we also recorded a triangle and claves for that reason. I could post those clips too if anyone cares." Quote: I suggest you redo the tracks with either white noise or pink noise ... run it at 0, -3 and -10db ... your approach to testing the frequency responce of different cards reveals nothing and has no basis in supporting any of your arguments. "This was not meant as a comprehensive test! It's just a reality check to counter the conventional wisdom that $25 sound cards suck by definition, and you'll never get pro results until you invest in expensive converters. As you can hear from these short clips, that simply is not true. Whether some people like it or not." I agree that many standard sound cards and even budget sound cards can do fairly well for recording. I know, I used them for many years and got better than average results. I'm a big advocate for beginners starting with their stock card and learning the basics and upgrade when they out grow it. Many of the 2 channel interfaces are no better than a standard sound card or may even be poorer. The only advantage to a 2 channel interface I see id a plug for a guitar or mic and a volume knob. Woop dee doo. No way in hell I'm going to pay $200 for that when I can do the exact same with a standard card and a preamp and some adaptor cords. I know better, Been there and done that. The card in my one computer is a stock Realtec thet will record or play back up to 24/196 and is a standard issue in computers now and does absolutely fine working with stereo wave files and good monitors. I would rather use 3 of those cards for a quarter the price of some 6 channel interfaces I've seen out there. Multi tracking is a different issue though. Back in the earley days bands were recorded in mono live and at the advent to stereo and three track could bounce and add tracks. The difference was engineers did the mixing prior to recording and would ride volume levels under live conditions basically the same way as a sound man runs mics during a live show. With multitrack cards the need for mixing prior to recording has/can be eliminated. In a case like mine I record 16 channels at a time under live conditions and capture the full frequency responce of whatever the mics produce. Other than having safe mic levels all the mixing is done afterwards. The shortcomings in both the preamps and converters of different interfaces quickly become apparent when mixing in the box. Manipulating the recorded signals to get a good balance reveals many things. Adding compression, EQing, Setting gain all take their toll on the original recorded signal and reveal the intrgrity and details of what was recorded. It also inlists the mixers skills to work with the shortcomings and identifying where ALL their bottelnecks may be occuring. This my friend is where a higher quality board can have an impact on a recording. I am granting there are just as many other factors as well including source, mics preamps etc. If theses suffer, obviously having a higher quality board is like pissing in the wind, you're going nowheres with it, this the expence of the board does not outweigh the rest of the equipment or skills needed to capture a good recording. Like everything else, as we make a decision to upgrade our recording chains we should identify where the biggest bang for the buck resides. Should we spend $200 or $300 per channel on an interface if there will be no great sonic improvement due to our front end defficiancies? I'll say this. If you plan to eventually achieve the highest quality you got to start somewheres. Its best to work from the front end of the chain starting with room, mics, preamps etc first because of the huge benifits That can be achieved from the investments being made. Then when you get to the interface as being the bottelneck upgrade. But I see nothing wrong with someone skipping the chain order and going straight for a high quality interface from the get go so long as the other components are eventually upgraded as well and they realise the upgrade may have no immediate benifits. If you're a wise shopper you should jump on a good deal when it comes along. Its one less thing to worry about later. Identifying the bottelneck can easily, and I mean real easily be misidentified by someone who doesnt KNOW where their problems exist. I see so many on this forum who Guess or take advice and have no fact established to identify where problems exist. This is where some on a forum can help others in focusing on their real problems, troubleshoot the causes, and give them their best advice to making improovements. Nathan your Test proves that an amature can easily misidentify one part of the recording chain and I'll give you a thumbs up for that. It doesnt go full circle though and show them how to identify the actual shortcomings of the cards and blurs an otherwise good excersize in troubleshooting someones recording chain. In that case I have to give you a thumbs down. I know your test was to invoke some discussion and you've done fairly well making your point against some of the responces, (Including some of mine which I hope you realise were designed to invoke a good debate here) Ill leave you with these items though Those who purchase the high end stuff, unless they are complete idiots, spend some time researching the various manufacturers, test the equipment for themselves, and KNOW what they are investing in. They have identified it properly through analysis and know its a bottelneck. Getting opinions on a forum is going to be very limited though. Those who own high end stuff, may have worked their way up through lesser systems over time as newer became available and know some pitfalls and shortcomings especially on some of the cheap crap out there people waste their money on thinking they'll get pro results. Comparing simular high end stuff may be impossible without biased opinions on what someone currently owns. There arent many out there that buy several high end systems and have no hands on with several systems to give a comparison. And those comparisons will have such a small difference that cost, maintainability, upgrede path etc all become a bigger deciding factor of what someone purchases. I sure wouldnt want to purchase a high end unit where the support sucked and I couldnt get fast repairs for my business for example. If I was spending $700 a channel I expect a vendor to bend over backwards to keep my business. Visiting many different manufacturers forums, reading their Q&A sites, Reviews, even this sites extensive product reviews can be a great source for making a decision in a purchase. Its more like checking for possible problems first to level the sales hype the adds give you. I would want to know a products failures, common problems and strengths etc before I make an investment (and should have on some purchases i've made in the past) Since the internet has been made available there alot of great info and direct answers to applications and work arounds people have access to and it should be utilized and contributed to.

Track 2 has ever so slightly reduced bass and trebble responce. Never having used an appogee nor the newer sound blaster its impossible to identify which is which. As far as I can hear the wire capacitence could account for the loss on the second track. The tonal quailites of a mellow acoustic guitar are a poor sources for an test as well. Niether frequency extreme is being challanged. I suggest you redo the tracks with either white noise or pink noise so theres a full frequency responce recorded and the full frequenct spectrum is being challanged. I also suggest you run it at 0, -3 and -10db the way cards are actually tested to reveal their strengths and weakneses. White or pink noise are the standard in the audio components and is whats used along with various sine, square, and sawthooth sweep generators to gauge the components specifications. Most sound cards can do fairly well in the lower and upper mid regions as in your test. Again much has to do with the conversion from digital to analog using multiple channels at different frequencies. You can download free test programs that generate these test frequencies, but using them within the same daw that contains the cards you're trying to test has is a problem if you use the converters in the test. Dont know if you have one but I suggest you get a high quality audio frequency test disk and use it as any competent audio engineer or audio technician would. You can play it back into your sound card from a high quality CD player and get the real story as to what your cards can and cannot do. Otherwise your approach to testing the frequency responce of different cards reveals nothing and has no basis in supporting any of your arguments.

"Hey look, a trump card for every a/b test ever". Thats a responce graph using a Microphone. Its not human ears with a mass between it that should be tuned to operate properly and have good audio depth perception under most conditions. "four inches apart in a 16 by 11 by 8 foot untreated room" I dont know anyone who tries to mix in a cave but I expect there may be exceptions. An interesting note though, tibeten monks have been able to use the complex acoustics of an untreated room to create absolutely astonishing effects just recently released on a recording. You also have the 2001 space oddesy soundtrack, Lux aeterna, for 16 unaccompanied voices by Gyorgy Ligeti and can hear how "Elmer" (as papa john phillips called it) appears sounding like an overdriven reverb or wierd string instrument created with multiple voices. Its based on using the voice to create natureal and unnatureal harmonic feedback from the room resonance itself to create notes that arent being directly sung by any one individual. So I guess untreated rooms can be used for some astoniching effects if used outside the box.

"Unfortunately that's just not true. Double blind tests are the gold standard in science for a good reason! Everyone is subject to bias, including seasoned pros. Even me and you". I'll see if I can listen to those samples some time this weekend. I have rehursal/recording with mu band after work, then I have a band coming in tomorrow to record demps, then I have to play out saturday night. I'f I'm not completely blown out by sunday I'll crank up my mastering computer and listen to them through decent monitors. In your A/B test you are only presenting the cards ability to record. Once its recorded you're posting it and relying on someone elses card to complete the full circle. This is only 50% of what a card must do for you albiet an important part. The playback portion of the card is just as important as the recording if its being used to mix with. The conversion back to analog may be enhanced, Flat, have different gain levels etc that are just as critical when mixing. So consider the test to be crude at best. In any case most of us are dealing with our own equipment which we know inside and out, literally. We now for some music under certain conditions its a piece of cake getting good results. In other cases trying to get transparency, the intermodulation is like having a root canal. In my case with two identical boards I can hear the difference between the two under certain circumstances especially with the drums. I'm not currently working as a pro any more but we all want to get the best resolution possible and not have to work any harder at it than we have to. I dont dispute any numbers you may have for what you consider to be a good cost per channel, but those of us in the business dont pay retail prices so again that cost varies greatly. I paid $6.25 per channel on used equipment where others would spend $40. As far as the sound blasters go, I used to use one for mastering my stuff years ago and I'm very familure with how well they work. I also have Yamaha cards, crystal, C-Media, Ess, M-Audio, Hercules and realtek. All of them sound different and have their pluses and minuses. Some of the differences are with their analog portions some of its their drivers/Gui and some of its the quality of their converters. For everyday listening or simple stuff they are perfectly fine. I'm surprised how many kids dont take advantage of using several cheap cards to multitrack until they learn the basics. But when the rubber meets the road and you're highly focused on mixing or mastering your equipment and skills using the equipment must just be vastly superior to stock stuff if you expect to have superior results working with raw takes and trying to make them sound professional. As I said those of us who have used pro gear, get quickley spoiled using it, and may be dealing with sub quality components and can do absolute magic with it in the eyes of an amature. Problem is we know the difference which is no placebo, its raw fact. Those extra few steps to achieve the highest fidelity possible that separate the amatures from the pros is an expensive gap to cross and I not only know why its there, but I see how its being eroded a littel more each year by better and better quality stuff being produced each year.

Ethan If you consider good enough to be good enough for what you do its absolutely fine with me. Some of us can actually hear the differences between live amd memorex quite distinctly will continue to strive for perfection. I dont know if the cost justifies some of the high end stuff. The prices on some stuff for what you're getting is a joke when you break things down to its parts, but its a small industry without a whole lot of competition. I know if I were rich I'd own the high end stuff but I opted to own real estate. I consider that Placebo thing you talk about to apply more to pot heads and amatures who really dont know what they're hearing or what they;re doing to begin with and does not apply to pros who have spent a lifetime doing this. Many who have a full background in audio including electronics degrees years do know the difference. Asking us to not notice its there is pretty rediculous. For those of us who know how it influences an entire composition know its the sum of the parts not the percentage of a single part that drives us to inprove our equipment in order to obtain higher transparency of the music we work with. In the mean time we deal with it best we can with it and use all the tricks in the box to work or around many different shortcomings.

Theres alot of truth and opinion in this thread. A/B ing a signal like Dan did does reveal some truth. His bandwidth is wider direct then it is converted. There may be several factors involved. Part of it may be the amplifying circuitry. Not all amplifying chips are equal and thats a fact. Part of it may be the converters or the clock. Part of it may be using the same board to feed both signals. Signals of different strength may produce a different frequency responce with an analog mixers volume or gain knobs set differently. I believe Dan can tell the difference between these as many others can. For an example you can even compare standard sound cards. I must have 6 or 8 of them around. Each have different amplifying circuits and can record at varying degrees of quality. They all play back at different qualities as well. I know when components are produced by different manufacturers the chemical makeup of the chips, impurities used to dope the silicon, and the etching of the chips is going to vary with different chips. Your best manufacturers are going to have the highest quality standards, highest rejection, best quality compounds, and the best engineered designs. These all break down to higher cost for higher quality. I cant go with all converters/clocks or preamps being the same. Anyone who has done any audio curcuitry building even on the most basic level would tell you some chips manufactured by one vendor can be vastly superior over others with the exact same specifications. Even flux residue left on a board can alter wave shapes. With the micro circuitry used on boards now all of these along with trace wire capacitence can compramise signal integrity. I guess what it comes down to peoples opinions of weather these combined items are going to be a big enough negative factor to influence their sound quality, and if having identified the influence, a person can or needs to compensate for any deperciation in quality. I admit these can be very minor items, maybe audiable to some and not to others. Some music may reveal it and other music its a non issue. To an accute audiophile, someone into recording, These kind of things can stick out like a sore thumb. Back when I did pro audio repair and could earn a decent living doing so I'd get people who would bring stuff in complaining of sound quality problems in one channel vs another. I'd compare both channels with a dual trace oscilloscope and a frequency generator and signal tracer to track the problem down. It could be just about anything influencing the two from voltage variations, components that were overdriven and lost some quality, heat, dirt, rosin, corrosion, or just a crappy design. Sometimes the unit tested perfect only to find the person had crappy wire or speaker variances. Others may have moved the system to a new room and it sounded different to them. In situations like that it requires fixing the customer and restoring confidence in the persons equipment. Whose to say that person didnt have a speaker blasting and ear at a concert or had fluid in their ear influencing their hearing. Still most experienced people who work in audio can rule out most of these items and deal with the rest either by compensating or masking the problem so it cant be heard. Their remaining skill is still hundreds of times more accute than a casual listener and can work wonders with the tools at hand.

"but when you get lost in the details, it's not worth doing anymore". Yes sir. I can take maybe 2~4 few hours mixing with these M audio cards and thats about it. I feel so fatigued having to concentrate so hard to get things properly mixed. I think part of it is the clocks are crap. Theres only so far you can go hearing the details as is and theres still wierd things happening to the sound. I can hear some kind of low level modulation happening between tracks thats very hard to hear but its there especially when I'm using alot of CPU. I thought it might have been something else but I been gradually ruling them all out. I'm seriously thinking about buying a dual clock to sync both cards. I was checking one out thats in my price range and it seems like it may be what I need My old card in comparison has a strong tight crispy high end. It also recorded tight sub lows much better. M-Audio chops them off below 30 I think. Its more likely 50 hz where sound starts sounding decent. Isis 8 track cards are very old now but they did have devcent converters for its time. It wont do anything over 16/48 and it wont run under XP but it does produce a darn tight sound when all is said and done. Just goes to show you newer isnt nessarily better.

Dan, I actually think its good you can bypass the converters and compare the two. You can actually compensate for some of the alteration in the process by goosing the source a bit. I went the M-audio route myself. I took a sound quality loss from my previous card soto get more channels. Spent $100 for 16 channels which was a good economy buy. I'm good enough to work with the losses to get reasonable results but I know the losses are there so it remains a challange with all I do. Theres no way I can afford what Fletch has to use in his business. (I'm just waiting till he dies and wills me the stuff). In any case the differences may be great from what I'm using but it doesnt stop anyone from creating great music. Even with its limitations an 8 track board can do wondorus things if creative minds are being used. It just sucks if you truely do record something great you'll have to bite the bullet and pay Fletch (or someone who has that kind of setup) to re-record the whole thing again to get that missing edge you cant get from some of the low end junk out there.

Good point on the Clocks there Fletch. Clocks can be provided by the card internally or externally. A good clock signal is a key to clean sample conversion.

Sound Quility is all front end stuff. Performance, Room, Mics, Preamps, Interface converters etc. Once the signal is digitized all is equal. The exception may be bus noise, latency, buffer size that may cause issues with crackle or timing of the data. Having a lower end DAW is not going to alter data like say analog signals would sound differently through a high or low end Hi Fi system. A higher end daw has indirect affects on the data. Higher processing speeds allow for more tracks recorded/mixed without dropout, Higher cpu consuming plugins can be used, Higher communication/sampeling rates, less latency etc. None of these are factors that are going to impact sound quality because the data is no longer an analog signal consisting of wave lengths and frequencies etc. It is a Digital audio broken down to binary math consisting of ones and zeros which is stored and recalled on to a medium like a hard drive or hard memory. Digital alteration by things like plugins would be the same between any box that has the memory and CPU speed to preform the program tasks a plugin program requires. Failures by plugins due to lack of resources consist of a crash, freeze, shutdown etc. They dont (Shouldnt) drop audio bits to keep up with the data processing. The data stream is being counted by the CPU as told by the program so nothing is lost. If some processing is performed and the results come up short there are several things that may occur. It may wait, it may re request the info to be sent again, it may get confused and hang, crash etc. Its not going to produce its own information to fill missing data thereby altering the sound on its own. The benifit of having a high end box is it can read the difference between an on and off bit faster and can multitask, communicate, and data stream data faster to keep up with the commands the CPU and programs are requesting.

There were tons of songs in the 60s that dealt with space directly or indirectly. Future, doubble meanings etc. Any thing went so long as it was well done. When you think of number of song like In the year 2525 done a few year later, Aquarious done on broadway, Rocket in my pocket, Venus, numerous others looking forward at a new world.

I think what he's explaining is he wants to use his laptop as a midi modual for his keyboard. Phils right about PCI cards as being the fastest for audio recording. Communications speeds are tied into the bus vs traveling through a Firewire or USB interface. But for midi, they make midi specific pci midi cards which achieve very low latency. Check with Blue2. He answerd a post with a board type he used to use that will do what you want. I cant remember what it was. Maybe find the post from a few weeks ago under recording forum.

Heres a vocal compression reverb trick to liven up vocals. http://www.recordinginstitute.com/R2KREQ/excomp.htm

Albietdamed {I have to point out some misinformation in this post. A cardiod mic does not lose all its low end beyond 6". It loses its proximity effect bass boost, but not all its low end. Lots of singers stand a foot or two back from a cardiod mic when recording, specifically to avoid the proximity effect. Others crowd the mic because they want the bass boost}. Loosing its bass boost or loosing its low end pretty much sounds the same to me. I think you're just parcing words. A standard cardoid like an SM57 will have its maximum frequency responce within the first 6". Beyond that the low frequency responce curve tapers off. My comparison was to point out the responce differences between a standard cardoid SM57 many were suggesting in comparison to a large diaphram. It was meant for a novice. I could have simply put it that a standard cardoid like an SM57 has a steep proximity curve that may be more difficult recording a folk guitar because a slight movement in distance from the mike will affect the low end greater than a large diaphram would.

He could also get a piezo mike installed under the bridge and record direct, then just focus on a high quality cardoid vocal mike. This would give the flexability to both record both on seperate tracks and be able to play out live at a later time. Maybe an SM7a or one of the Betaa like a 52 or 58 might be decent as well.