44.1 vs. 48 Sample Rate

[andrewd]

OK simple question for y'all. I don't wanna get to far into our new record and then realize I screwed up.

I have done all of my recordings up until this point at 44.1. Yet, my band has started our new album at 48. When I bounce (Logic Pro) things down after mixing, will I get ANY problems... tempo changes, distortion, overall wiredness, etc?

I have done recordings and burned them down to MP3's at 48 and they sounded great, that lead me to starting our record at 48 also. I've also heard its easier to mix at 48 and then bounce down to 44.1.

Any help would be awesome.

D

[Whitecat]

Bear in mind that Logic's sample rate conversion isn't the best of what's around.

http://src.infinitewave.ca/

You would get better results using a different program to convert the files to 44.1.

[opium89]

Em, pretty graphs! What is their significance?

[matt thomas]

make sure you convert the files as opposed to get logic to play the same files at the new sample rate, in which case you will get a change in tempo and pitch

narco

[optix]

Originally Posted by terminal3

Bear in mind that Logic's sample rate conversion isn't the best of what's around.

SRC Comparisons

You would get better results using a different program to convert the files to 44.1.

That's really interesting terminal3, thanks for that!

(opium89, as I understand it the first graphs that come up show a sweep which was recorded at 96kHz and then downsampled to 44.1kHz; you can see how various programs introduce background noise etc.)

[Bob Olhsson]

I wish there was a simple answer.

All but the most expensive A to D converters are compromised at 44.1 so in many cases 48 will make the end product sound better.

Unfortunately using less than the best SRC is also a problem.

About all you can do is try both ways and listen. I'd probably use 48 and then make 320k MP3's at 48 for listening.

[Recording David]

I think the sound difference between 44 and 48 is absolutely negligible. 48 is the standard in pictures, so I wonder why music presumably intended for CD needs to be recorded at 48?

Sure, go for it if you're going to record at hi rates (88 upwards).



Anyway, if it's all going at 48 get a good SRC - if you're on a budget have a look at this: Audiofile Engineering

It's fantastic, on a par with Barbabatch if you ask me.

[theblue1]

Bob, I'm guessing you're referencing filtering issues with regard to "all but the most expensive" converters operating poorly at 44.1 kHz... I'm thinking it's also often filter issues that keep the SRC of lesser-tier soft and hardware from delivering artifact free conversion.

Am I tracking your thinking correctly?

[opium89]

So is the background noise represented by color in the background? Anytime and Digital Performer bad, Adobe Audition and Sonar good?

[theblue1]

Sonar 6 SRC I believe showed an improvement over previous versions in those tests linked above.

But if you look at the results for Voxengo's free R8Brain SRC app, you may note that the somewhat aggressive filtering at the high end looks awfully similar. (Voxengo has supplied Sonar with some other technologies. I couldn't help but wonder if this wasn't one of them.)

[Brian M. Boykin]

Just some things to think about:

1) If your going to have it mastered, send it in at 48. Let the mastering engineer down sample it to 41. He should have a better src.

2) If you and your band have the cash and you have a techi member, think about a two track analog deck. You can mix to it at 48 and then re-record back at 41/16.

I bought an otari mtr-10 2 years ago. I payed $299.00 dollars for it, had JFR optically align the heads and payed a $100.00 for an alignment tape. I have less than $900.00 invested and it is rock solid. I picked it up because at the time I started building my studio most studio's, and I called studio's from the east coast to the west coast, seemed to agree that they did not trust algorithim's for down sampling their projects. I will admit that it seems that in the past year or so that a lot of people seem to think that is not the case anymore, but it's a nice tool to have if you want to take the plunge. If your not willing to set the deck up right though, don't do it, it won't perform for you and will only leave you frustrated. I will also say, don't let analog scare you, it's not as difficult to understand how to set one up as it is sometimes made out to be.

3) I also record at 48 because of the filtering issues that seem to have been raised in earlier posts. At some point a converter has to roll off the high end to prevent alaising. At 44 it can start in the audible hearing range. 48 hopefully starts the roll off outside of the audible range. That's the theory at least.


Good luck,

Brian

[8th_note]

Thanks, terminal3, for that link. I think it may have explained a mystery for me.

When I got my converter I wanted to see how much improvement I would get at 96 compared to 44.1. I recorded a band (one song) using a total of 7 tracks (4 on drums, 2 guitars, bass), overdubbed, once at 44.1 and again at 96. I mixed the tracks down to 44.1 without any eq or plugins and compared the results. The 44.1 song sounded better - and it wasn't subtle. The 44.1 sounded cleaner, clearer, and had better dynamics. I've been recording at 44.1 ever since.

I would guess that the SRC in my software was the cause of the difference. I may try that again using Ozone to see if the results are different.

[theblue1]

Originally Posted by Brian M. Boykin

Just some things to think about:

1) If your going to have it mastered, send it in at 48. Let the mastering engineer down sample it to 41. He should have a better src.

2) If you and your band have the cash and you have a techi member, think about a two track analog deck. You can mix to it at 48 and then re-record back at 41/16.

I bought an otari mtr-10 2 years ago. I payed $299.00 dollars for it, had JFR optically align the heads and payed a $100.00 for an alignment tape. I have less than $900.00 invested and it is rock solid. I picked it up because at the time I started building my studio most studio's, and I called studio's from the east coast to the west coast, seemed to agree that they did not trust algorithim's for down sampling their projects. I will admit that it seems that in the past year or so that a lot of people seem to think that is not the case anymore, but it's a nice tool to have if you want to take the plunge. If your not willing to set the deck up right though, don't do it, it won't perform for you and will only leave you frustrated. I will also say, don't let analog scare you, it's not as difficult to understand how to set one up as it is sometimes made out to be.

3) I also record at 48 because of the filtering issues that seem to have been raised in earlier posts. At some point a converter has to roll off the high end to prevent alaising. At 44 it can start in the audible hearing range. 48 hopefully starts the roll off outside of the audible range. That's the theory at least.


Good luck,

Brian

bold added


If you're a young dog.

But, yes, some young people and the exceptionally rare adult can hear above the 20 kHz afforded by good 44.1 kHz conversion/alias filtering. But it is pretty rare for adults to hear much above 15 kHz. (When I was a kid 16 kHz was often cited as the typical top end of human hearing.)

And there is no empirical evidence or even credible hypothosis that I know of that humans have some kind of perception of audio above the range that the ear and ancillary nerve systems respond to, although, expectedly, there is evidence of nerve activity around the threshold of reported audibility... (the "I might here something" range, as it were).

[Brian M. Boykin]

You're right theblue1,

I did say " That's the theory at least". I have checked my ears with my tone generator and I can hear to about 16k, everything after that makes my ears feel a sensation, but I can't say that I can make out any sound. 48k gives a little extra headroom. As the theory goes, alaising can show up in th 10k range, I would definately hear that. I would rather be safe than sorry, and mixing to tape is not really a bad thing to do as long as the deck is set up right. Generally speaking, I would not record at 48k and the have the computer down sample it to 44.1k. I would however record at 88.2 and then drop to 44.1, as it is just simple division as far as plotting samples. I will admit that even that is being a little anal retentive. If I were him I would just record at 44.1; bit debth is another can of worms.

Brian

[theblue1]

Yeah the synchronous sample rate (double or quadruple rate pairs, as it were) thing seemed to make so much sense to me -- and according to Dan Lavry it's easier for the SRC to get a clean downsample to a synchronous rate -- but, also according to Lavry, there's nothing to stop a good SRC from going from, say, 96 or even 48 down to 44.1 with full accuracy in the audible range. In fact, I think Bob Ohlsson and Lavry straightened me out on that in these very forums. (I was walking away going, well, who does this Lavry guy think he is, anyhow -- then I found out it just who he was, did some more homework and... stood corrected. I don't mind admitting I'm wrong when I'm pinned down by an industry legend or two... )


Anyhow, I think whatever gets one the sound he likes is the right answer at any given time. I do think it's interesting -- and ultimately quite helpful -- learning about the technology and science behind what we do.

In the past, I'll admit I was too eager to get on with it to bother that much with book learnin'... but I like to think I've taken advantage of the internet to learn a lot of what I should have learned in school -- heaven knows at least a couple people were trying to drill it into me.



One thing that's stuck in my head -- and always made a lot of sense -- I've ready Lavry and others suggest that while 44.1 kHz as a standard sample rate forces failry steep anti-alias filtering to deliver the target 20 kHz bandwidth and anything higher than 96 kHz is not just unnecessary but tends to produce alias error in audible bands that must be mathematically corrected out, that something in the 60-70 kHz sampling rate range would give a frequency bandwidth well surpassing the highest recorded human perception and afford plenty of room for gentle filter slope.

[Brian M. Boykin]

Yeah,

I read all those threads when they were posting them. It must have been a couple of years ago now. I was day dreaming about 192hz at the time, that's when 192 was just starting to roll out, and I was looking for converters with multiple ins and outs. The phrase that I remember most is "It only takes two points to sample a frequency." Once I got what they were saying I decided I did not need 96 on up, so I looked for converters that would fit my needs. I went with a motu 2408mkii and 1224's. I recorded at 44 until I got my deck and then I went to 48. Motu makes mid-range stuff, so knowing that, 48 gives me peice of mind that nothing screwy is going on up stairs, especially since I mix to tape.

Brian

[AllAboutTone]

To me the bit rate is way more important than 44.1 and 48, im not a 48 fan at all. 88.2 is kewl sometimes. For a different vide or artist i have go to 32 for fun.

[Lagerfeldt]

Avoiding SRC is better than recording in 48 kHz going down to 44.1 kHz, even using a good SRC, IMO. Most good A/D's have a decent filter and the compromise is usually a theoretical matter. My hearing goes up to about 18 kHz, after that it fades rapidly. Mind you, you need to good monitors and playback to check.

Personally I use Barbabatch for SRC.

As someone said above, recording in 24 bits is way more important. 24/44.1 kHz is what I use.

[Dave Derr]

There is some evidence that humans can percieve implied frequencies above 20KHz in the form of transients. While tests have shown we cannot readily percieve steady state tones above 20Khz, scientists are researching some outer ear nerve hairs, which seem to respond to sharp rising edges.... Transients.

The closer a transient event is to us, the sharper the rising edge is. This could give Darwinists cause to believe that our species had an edge in surviving when they could tell a rock hit 2 inches from our heads, instead of 10 feet. Since the ability to detect and distinguish the "sharpness" of transients (which have a theoretical infinite frequency response) gave an edge to those who had it, this trait could become dominant over hundreds of thousands of years since the humans with this ability lived and procreated longer on the average.

While perhaps we DO NOT hear steady state tones above 20K, there is a real possibility that we can tell when a transient has been slowed up and does not contain frequencies above 20Khz.

Two other things to think about:
1) Most of our speakers do poorly above 20Khz, so perhaps in most cases not having 23KHz in the digital sampling system isnt much of a bottleneck.
2) The digital filters used in our current converters produce a "pre-echo" to a transient. If our ears are truly transient sensitive little buggers, this pre-echo could really throw them for a loop, since a pre-echo on a transient edge is nothing that ever happens in nature. Hutch from Manley Labs and I have had many conversations about this FIR filter-induced anomaly.

[peeder]

Hi Mr Derr,

We've discussed this at some length here recently. The conclusions seemed to be that those ultrasonic transients actually have plenty of low-frequency information in them and that is what we are hearing. If you filter the low-frequency out, you can't hear them anymore, and they don't look like transients anymore.

The rise time of amplifiers and speakers in practice isn't fast enough to capture ultrasonic rise times so the angle of a leading edge of a transient is going to be softened anyway in real life. I think you can get to "maximum" within the audible band.

My conclusion is you can't hear >20k and it doesn't make a difference anyway. However, the audible-band artifacts from ultrasonic sources, such as intermodulation distortion, can be heard, although they usually are already captured in the process of recording and preserving the ultrasonic information that generated them will simply double them unnaturally upon playback.

Now to the OP, as to 44.1 vs. 48, there is no substitute to testing your specific implementations of ADC, DAC, DSP, and SRC. I suggest not relying solely on subjective listening tests but instead using nulling, test tones & spectrum analyzers/spectragrams etc. and don't ignore low-frequency signals even though many will argue the high frequencies are where the differences are more apparent.

If we had a Consumer Reports-style independent testing lab we'd just be able to look up the performance characteristics of our gear and not need to do this testing.

Originally Posted by Dave Derr

There is some evidence that humans can percieve implied frequencies above 20KHz in the form of transients. While tests have shown we cannot readily percieve steady state tones above 20Khz, scientists are researching some outer ear nerve hairs, which seem to respond to sharp rising edges.... Transients.

The closer a transient event is to us, the sharper the rising edge is. This could give Darwinists cause to believe that our species had an edge in surviving when they could tell a rock hit 2 inches from our heads, instead of 10 feet. Since the ability to detect and distinguish the "sharpness" of transients (which have a theoretical infinite frequency response) gave an edge to those who had it, this trait could become dominant over hundreds of thousands of years since the humans with this ability lived and procreated longer on the average.

While perhaps we DO NOT hear steady state tones above 20K, there is a real possibility that we can tell when a transient has been slowed up and does not contain frequencies above 20Khz.

Two other things to think about:
1) Most of our speakers do poorly above 20Khz, so perhaps in most cases not having 23KHz in the digital sampling system isnt much of a bottleneck.
2) The digital filters used in our current converters produce a "pre-echo" to a transient. If our ears are truly transient sensitive little buggers, this pre-echo could really throw them for a loop, since a pre-echo on a transient edge is nothing that ever happens in nature. Hutch from Manley Labs and I have had many conversations about this FIR filter-induced anomaly.

[theblue1]

Originally Posted by Dave Derr

There is some evidence that humans can percieve implied frequencies above 20KHz in the form of transients. While tests have shown we cannot readily percieve steady state tones above 20Khz, scientists are researching some outer ear nerve hairs, which seem to respond to sharp rising edges.... Transients.

The closer a transient event is to us, the sharper the rising edge is. This could give Darwinists cause to believe that our species had an edge in surviving when they could tell a rock hit 2 inches from our heads, instead of 10 feet. Since the ability to detect and distinguish the "sharpness" of transients (which have a theoretical infinite frequency response) gave an edge to those who had it, this trait could become dominant over hundreds of thousands of years since the humans with this ability lived and procreated longer on the average.

While perhaps we DO NOT hear steady state tones above 20K, there is a real possibility that we can tell when a transient has been slowed up and does not contain frequencies above 20Khz.

Two other things to think about:
1) Most of our speakers do poorly above 20Khz, so perhaps in most cases not having 23KHz in the digital sampling system isnt much of a bottleneck.
2) The digital filters used in our current converters produce a "pre-echo" to a transient. If our ears are truly transient sensitive little buggers, this pre-echo could really throw them for a loop, since a pre-echo on a transient edge is nothing that ever happens in nature. Hutch from Manley Labs and I have had many conversations about this FIR filter-induced anomaly.

Here are a couple of widely recognized, highly trained experts on the notion of special properties of high frequency transients:

Dan Lavry writes about this audiophile-promoted idea that HF transients are somehow a special case and perceivable above the normally accepted frequency range of human perception:

The fact is: ALL the music is transients, NONE of the music is steady state. What is a transient? What is steady state? I already explained earlier that steady state is a "forever" repeatable signal, where each cycle is the same as the other. Where in music do you have such a thing?

So perhaps one can talk about a "semi periodic" portion of a signal, where you have a "bunch of cycles" that are near repetitive for a while. Even a piccolo's highest note is not higher than say 4.2KHz or so. A steady as possible high piccolo note that lasts 1 second is limited to about 4000 cycles. But the attack time and decay time which the time when there is much "change in waveform" is the "opposite of repetitive", it is a almost "totally transient portion of the signal"...

What about a really short duration sound? A drum? What about a piano? After the attack, the amplitude drops down which makes it less "periodic like"... So at the end of the day, transient means time varying in a non periodic manner.

With all the audio hype, the "street talk" refers to transient as some short duration high frequency thing, but it is not so by definition. But even if I am willing to "bend" with the "flow" and call a drum transient, it is a mistake to say that the quickness of it all frees it from lower frequency content. In fact, the opposite is true, like the simple example of a sudden step where you add DC (very low frequency content). The addition of DC happens fast, and that part of the energy is both small, and contains higher frequencies, but the bulk is at low frequencies. The frequency plot of a step energy distribution is 1/f...

Lavry continued:

All that talk about high frequency transients assumes that one can have a signal activity at very high frequencies (above hearing) that does not involve low frequency activity. The only way you can have such a signal is to insist that it is periodic (lasts forever) with all the harmonics at high frequency.

We all agree that there is a high frequency limit of hearing for such a signal, a periodic one. But some "less technical" folks and all sorts of sales material also suggest that we respond to high frequency transients (non steady state).

The fact is: a non steady state high frequency is always accompanied by low frequency energy. A gated sine wave say 10 cycles of 30KHz sine wave has energy at 30KHz but also at low frequency, due to the gate. Let's take the simplest example: A single fast step from zero to 1 volt. True, the fast step requires a lot of high frequencies, but do not forget the change in DC (DC is the lowest frequency - 0Hz). In fact, a Fourier integral (NOT TO BE CONFUSED with Fourier series) will show some spread of energy into audible frequencies.

So all the reports about hearing high frequency transients improperly suggest that the test tones (or music) consists of only high frequencies. In fact, there is always audible energy involved, and the listener responds to that energy while thinking they respond to the high.

PSW Recording Forums: Dan Lavry => the "high frequency transients" fallacy

Ethan Winer, in Audio Media Magazine (UK) on the general topic:

Myth: Even though people cannot hear frequencies above 20 KHz, it is important that audio equipment be able to reproduce higher frequencies to maintain clarity.

Fact: There is no evidence that a frequency response beyond what humans can hear is audible or useful. It is true that good amplifier designs generally have a frequency response well beyond the limits of hearing, and the lack of an extended response can be a give-away that the amplifier is deficient in some other areas. If for no other reason, though there certainly are other reasons, an amplifier's effective cut-off frequency - the point at which its output has dropped by 3 dB - must be high enough that the response loss at 20 KHz is still well under 1 dB.

With audio transducers - microphones and speakers - the frequency beyond which they do not respond (the cut-off frequency) is often accompanied by a resonant peak, which can add ringing and a boost in level at that frequency. Therefore, designing a transducer to respond beyond 20 KHz is useful because it pushes any inherent resonance past audibility. This is one important advantage of those expensive condenser microphones that use a tiny (less than 1/2-inch) diaphragm and are designed for critical audio testing.

It is very easy to determine, once and for all, if a response beyond 20 KHz makes a difference. All you need is a sweepable audio low-pass filter. You start with the filter set to well beyond 20 KHz, play the audio source material of your choice - I've used a set of keys jingling in front of a high-quality, small-diaphragm condenser mike - and sweep the filter downward until you can hear a difference. Then read the frequency noted on the dial.

Dispelling Popular Audio Myths

[max cooper]

Originally Posted by Dave Derr

There is some evidence that humans can percieve implied frequencies above 20KHz in the form of transients. While tests have shown we cannot readily percieve steady state tones above 20Khz, scientists are researching some outer ear nerve hairs, which seem to respond to sharp rising edges.... Transients.

There was a good article somewhere that made the point that increasing the sample rate gives the most improvement in the low freqs. Anyone remember that one?

[Jayro_Rockola]

"The frequencies we cannot hear effect the frequencies we can..."
-Wise old audiologist (not me)

[jude]

so why not just go back to tape?


ohhh, thats right, then forums would be empty. people might start recording again. **** sample rates and bit depths. good music is what the paying consumer wants to here. not some bullshit "well this was recorded at 96k, so its better"

most consumers these days dont know/care.

who here can honestly say they can here the difference between 44.1 and 96k when its converted to mp3 and played thru 1Pod headphones, huh?

[Brian M. Boykin]

If I'm following Dave right, the first part of his post suggests we hear frequencies that were not in the original source material because our brain inserts them. I'm sure I read somewhere that in the lower spectrum if the brain hears harmonics of a lower frequency it will insert the lower frequency even though it does not exist in the source material.

Lavry sounds like he is saying music is made up of transients; notes are not steady state tones. Upon looking up the word transient my webster's dictionary,it says, "passing away with time;temporary, passing quickly, temporary, passing quickly; fleeting, staying only a short time." So, the beginning of a piccolo note is a transient by definition and so is the decay as well as the peak. My understanding until now, as a drummer, is that the spike is a transient, you know, the part of the note that throws your digital converter into digital distortion. Interesting new understanding I've obtained.

I guess Ethan has never posted results of such a test. That would settle it, if you percieve change in the source material and the filter is set at say 22khz, then you could conclude that frequencies above 22khz somehow change what you hear. I would venture to say not because you actually hear the frequency but because in the world of physics they somehow affect the frequencies you can hear. If that's true, then it really is a lost cause trying to control the effects of higher frequencies. They're everywhere, like uv light.

The pre-echo thing is also interesting, would like to know more about that.

Hey Max, my whole concept of how digital recording worked was completely screwed up for years because of an artical I read in a magazine about that very thing. It claimed that higher sample rates were better for lower frequencies because lower frequencies were more complex and required more sampling points to accurately recreate them. That does not make sence in that it presumed that all 96k frequencies were applied to one frequency. A frequency is called a frequency because it cycles across an axis a set number of times in a given period of time. So the converter only has to record that it exists or does not exist at a given point in time. Two point's is all that is required to do that. Is that what you were referencing.

As for how this has changed the way I'm going to record, if Dave's right then 48k will help my cause, can't record at 96k anyway. If Lavry and Ethan are right, then recording at 48k won't hurt, and my Otari mtr-10 is a great src and zero attack compressor that smashes all those peaks into tape saturated bliss. I feel pretty good for now, at least until the next post when someone changes my understanding, AGAIN!!!!!

Brian

[kafka]

Originally Posted by Brian M. Boykin

Hey Max, my whole concept of how digital recording worked was completely screwed up for years because of an artical I read in a magazine about that very thing. It claimed that higher sample rates were better for lower frequencies because lower frequencies were more complex and required more sampling points to accurately recreate them.

LOL! I guess we'd have to call this one the "Nyquist-Got-It-Totally-Backwards Theorem". "lower frequencies are more complex" That Fourier guy must have been an idiot! Imagine wasting your time breaking things down into piddly little sine waves, when he could have been working with "more complex" lower frequencies instead.

I'm glad you were able to overcome the influence of that article! And, please, whatever you do, don't provide a reference, or we'll all have to endure everyone without an EE degree quoting it for years.

[taturana]

Originally Posted by jude

who here can honestly say they can here the difference between 44.1 and 96k when its converted to mp3 and played thru 1Pod headphones, huh?

well... that is a pretty crappy argument that gets repeated over and over...

so since people are listening in mp3 quality does not matter? of course it does, and the best quality you record the better your mp3 will be...

not to mention that with increases in bw and mp3 quality these mp3 files will become better everyday, and people who actually believe that kind of crap will end up throwing away all their work, since it won´t be good enough.

i go for the best quality sound always. and that way it will sound better whichever media format is chosen.

and yes... let´s go back to tape.. that´s pretty damm nice. nothing like mixing to a nice 1/2" tape.. mixes just glue together real nice... and it sounds so cool... too bad we have to digitalize it afterwards..

[joeq]

a recent project of mine had songs that were recorded at 44.1, 48 and 96. each song was mixed at its native sample rate and all were mastered at a place that uses an analog chain- out at the native rate and back in at 44.1 with top-notch converters

the differences between songs due to song structure, instrumentation, players, mixes, etc totally obscured whatever sonic differences between the different sample rates that might have existed. Certainly none of the tunes jumps out, waves its arms and shouts "higher sample rate!" In fact none of them even whispers "higher sample rate".

Nowadays I have to consult my logbook or the original session files to remember which was which.

[Dave Derr]

I totally agree that folks that agonize over the differences of 44.1, 48KHz, 96, 16 bit, 20bit etc should probably go back and learn what a good song is, learn what 1 KHz EQ boost sounds like, learn when a vocal is exciting, learn when the cymbals are too loud, learn when theres a tempo problem....

Really, the differences between the usual different sample rates and bit depths is just miniscule... and double blind AB/X testing has often shown that people can't even hear the differences reliably.

The scariest AB/X test was back shortly after 1990, when HIGHLY RESPECTED ENGINEERS, could not tell the difference between a live source, a single generation 16bit 48K ADAT recording, and a 4th (or more) generation ADAT copy, when done in blind testing! Everyone in the test was surprised... and embarassed.

Just make sure you have good syncing without digital error problems, and get on with recording music, I say! It's not the airplane.... it's the PILOT!. Its not the paintbrush, its the painter. It's not the drill, it's the dentist.... errr wait, bad analogy....

[Brian M. Boykin]

Ya Kafka,

And to make things worse, a friend of mine introduced me to a friend of his who explained to me the exact same concept, further reinforcing it. When I finally got the truth, about 3 years ago, I took on the attitude of learning what I could, understanding all the theories that are out there, and building a system within my budget that would be a compromise of what I thought was relative. Also, I WILL own a 2" deck one day. I'm going to pull it apart like it was a classic car and rebuild it from the ground up. In the end it will be a tool, a tool I say, to acheive a particular sound. I'm edging towards an mci jh16 or ampex mm1200, with all the color an analog deck provides. But I'm sure I'm techi enough to do that.

Brian