I would definitely like to see 24/96 Output Sound Quality so it can be used for studio use. Some people get masters from artists, and and it would be nice to try to keep the digital files somewhat on par with the vinyl records. I imagine also people who record their vinyl to the computer would also like this feature.

Well, that would require a whole new hardware (which isn't planned for anytime soon). USB1.1 wouldn't be able to handle 4 Ins/Outs at 96/24 (96000*24*2*4=184320000bits=184Mbits >> 11Mbits=USB1.1) You'd need firewire.

I dont think so. Read up the dynamic range of vinyls, and you will see, whether 16bit/44.1kHz or 24bit/96kHz wont make a differenc. Also compare this with the specs of your stylus needles ;)

actually, i've recorded vinyl at 16/44.1 and at 24/96 and compared and there is a clear difference when you have the right speakers or headphones. do it with the same song, then play back both recordings simultaneously, and then cut back and forth. that's all you need to do to know that it makes a difference. just becasue the frequency of the needle only goes up to around 20, doesn't mean that having four samples per full cycle isn't better than only having two. it's less choppy and more accurate. but i did imagine they would need new hardware and so that was what i was suggesting, i figure one day they'll upgrade. i guess it's good news that firewire could handle it, thanks for the calculated info.

Technically correct. Question is whether you can hear the diff', I cant, but as you can, 24bit makes sense for you. :-)

I think the 16 bit vs 24bit makes no difference, the 44.1k vs 96k will make a difference.

The dynamic range of vinyl records is well below that of 16-bit digital, a fact that anynoe who records records into a computer can easily verify for themselves (simply measure the difference in dB between the surface noise and the highest peaks). 40-50dB is pretty typical, equating to more or less 8-bit performance.

Most records are low-ass-filtered at the cutting stage pretty steeply around 18kHz. This is done to protect the cutting head, which can overheat and self-destruct from too much high-frequency material. Again, this is well within the limitations from 44.1 kHz sample rates.

The whole business of "more samples per cycle" being somehow better was debunked nearly 80 years ago by the Nyquist theorum: en.wikipedia.org

Heh, there are definitely some "low-assed" records out there, but what I should have typed was "low-pass".

Hm, besides the n-theorem, wouldnt it make sense, lets assuem we have a sine wave at 22.05kHz, with 44.1kHz sampling freq you would at best get only the peaks, if you have bad luck you might end up with sampling only the zero's. So either way with more kHz wouldnt you get a better representation of the sinewave with all its round edges etc.?

The relatively low dynamic range of vinyl is not a particularly good argument against going to 24-bit for digital recording.
I'd probably be more likely to do that then increase the sample rate - but I don't do either because I can't be bothered with the extra sample rate conversion or dithering to get to cd standard.
Some good reading and additional links here
www.soundonsound.com

Not once it gets converted back to analog, no...despite what you may see on-screen ni a DAW. Nyquist theory states that 2x the maximum frequency of interest is enough to use as a sample-rate, and anyone with a signal generator, A-D and D-A and an analog scope can easily verify it for themselves.

Sorry, I dont understand your post. I know you can reproduce the frequency itself with a little bit of luck (read above why I think it still can happen you end up with a flatline), but is this a correct representation of the original signal?

nyquist's theory is saying that the maximum frequency you can record is half the sample rate, it's not saying that you only need a sample rate double that of your highest frequency to get a perfect recording.

If that were true you wouldn't get studios recording at 192kHz since the human ear can only hear 20kHz at best.

Clear now. Thanks. Thats why I was surprised, so that means the n-theorem is quite useless in this discussion and this is wrong then:

24bit doesn't have any more of a dynamic range than 16bit (that's my understanding), it just creates more precise representation of any given dynamic range. and so that is why a lot of people in studios up-convert given audio from 16bit to 24bit before applying signal processing. i personally would rather have floor noise and lower dynamic range and better sound quality, than a super high dynamic range. i honestly can't say whether it is the bit rate or the sample rate that makes it sound better, i just know it sounds better, but more importantly, it hurts my ears less. the effect is that the sound is more surrounding, and coming less from a specific point in space. especially with vocals, but then mics only record at a maximum frequency, so who knows. but really for me, i would like rane to update to a 24/96 interface because everybody is recording their songs at 24/96 and higher. THAT IS MY SUGGESTION TO RANE. i think they should do it for the few people who care, and the masses that don't care can choose to continue outputing the sound at 16/44.1. as long as it works, i don't see any disadvantage of having 24/96, as long as people still had the option of outputting at 16/44.1 if they wanted to.

also, even if you had a 44.1 file, and you moved the record forward twice as fast, then the sample rate would in effect now be 88.2. so it seems that theoretically, higher output would also improve scratch sounds.

a'right, i take that back, 24bit does supposedly have a higher dynamic range. but here are some quotes i found online:

The 16-bit word offers 65,536 distinct dynamic levels.
The 24-bit word offers 16,777,216 distinct dynamic levels.

(which would explain why you would have less distortion of a 16bit file by converting it to 24bit before changing the audio levels in processing)

the human ear can hear time differences less than 1/44100th of a second

(which would be another reason to have 96khz sample rate rather than just improving the wave form representation of a 20hz frequency and might explain why 44.1 is harsh on the ears)

definitely, overall, i think having fewer steps and smoother lines can only be easier on the ears and allow for longer listening before suffering from audio fatigue.

and make it sound more like analog.

i think it is undisputed by anybody in the music world that vinyl records sound better and are easier on the ears than digital files, despite having frequency limitations and dynamic range limitations.

It's a misconception that higher sample rates make the line any smoother - it just allows you to represent higher frequencies (which we can't hear anyway!).
There are other advantages to higher sample rates, such as easier design of the associated digital low-pass filters - but somehow making the reconstruction more 'accurate' is not one of them AFAIK.
Bob Katz has a really good chapter about this in his book 'Mastering Audio - The Art & The Science'.

Higher bandwidth -> more strain for your pc, plus it will be more expensive. Why pay for a feature which only 0.1% of the users would benefit from? 99% use mp3s audio while using SSL.


That is due to rounding errors which get accumulated over the time and are worse if you sample down to 16bit again and again.


Deft, can you explain why? This is in contrast to what nobs said, and it doesnt seem to be true for me. Example: You got a sine wave near to the frequency limit, due to the n-theorem you can reproduce that frequency, but it will end as a triangle wave form (assuming you sample the peaks) and not as a round shaped sine wave. Can you enlighten me?

When it goes through the D-A convertors, it will come out (in the analog domain) as a "round shaped" sine wave.

Provided you stick to the Nyquist rule of greater than twice the cycle frequency, there is no 'loss' of information during the digital sampling process - provided that 'effective' filtering is in place to allow this.
I'm not an expert in digital audio sampling processes, but to say higher sample rates make anything smoother is the biggest repeated misunderstanding with digital audio I reckon. Sampling at 44.1Khz means you can totally accurately reconstruct any waveform at less than half the sampling rate.
I will try and find some links that explain it better than me - but you need to really look into the role of anti-alias and reconstruction filters to understand it more fully.

Alright if it comes out as a round shaped sine wave, then my original source is a triangle wave. The point is, during sampling there is no chance to distinguish between a triangle wave and a round shaped sine wave. Correct?

IIRC the anti-alias fiter means that it isn't a triangle wave being sampled though. I don't want to say too much for fear of giving you incorrect info - the filtering steps are crucial, and are very much a key part of the process.

IMHO any static filter will alter either the sampled signal or/and the output signal, so you cant have a precise representation of the original signal.

Anti aliasing on the output will have to alias based on the info it has, as it only sees the peaks from the signal I was talking about, it has to guess, it might alias the signal in such a way that you get a round sine. But then again what happens with a triangle signal?

These are my _assumptions_, not facts. The only obvious fact is, that once you got a sampled signal (and no further info) you cant say whether the orignal signal was a triangle wave or a sine way.

If anyone has any further info here please share it :)

O.k. I have found a good thread over at the Sound On Sound forums that goes over this very same ground.
Hugh Robjohns and others explain it far better than I could
www.soundonsound.com

This is an excellent ressource!

Okay, read through it Deft, and these are the key points:


My triangualar wave has high frequency components and low frequency components. High freq. comp. > half sample rate will be discarded, due to the alias filter/reconstruction filter.


Thanks for the links Deft, I think I got it now. :)

Cool - I didn't want to try and explain it and get it wrong! I just accept it as truth and try not to think too hard about it.

thanks for the info. i think that is true, that cd players take the points and then connect them with a smooth curve. to be honest, this all went over my head. but if rane is reading this, i would be willing to pay a whole lot more to get 24/96 output. all i know is that it does sound different, and that is what everybody is recording at. i literally can't listen to cds because i have ringing in my ears, and cds make the ringing a whole lot louder. mp3s are completely unbearable. no matter how low the volume is, it causes severe ringing in my ears. most vinyl does not affect my ears at all, though some does, and audio at 24/96 has less of an effect than cds. whatever the reason, that is the reality of my situation, so even if it cost $2000, i would be down to pay for it. i would be interested why final scratch chose to go to 24/96, and why studios record at 24/96 or even 24/192, if it makes no difference. i think it would make sense to keep things as close to the master as possible throughout the whole process since Rane is considered to be "professional" audio equipment. i assume that if Rane came out with a 24/96 interface, the 16/44.1 would still exist, and so people who don't have the money and who don't care could buy that one instead. pass the cost on to the people who want 24/96. and while many do listen to mp3s, there is definitely a large percent of people who don't listen to mp3s at all. and very few deejays that i know are willing to ever play mp3s in a club. straight up, weed smokers trip off the little details. also, i wish i had the link, but they did a study, mapping people's brain signals while listening to audio, and they found that high frequencies that people can't hear do interact with brain activity. they used special equipment to record sounds out in the jungle where there were naturally occuring very high frequencies. i just mentioned the difference in recording vinyl to the computer at 16/44.1 and 24/96 cuz i noticed a difference, but i actually have no intention of doing that, since i'll just listen to the records instead. i just thought that if one were going to give in and use digital files, that any benefits they might have could be appreciated. here is a discussion i found on the subject:

studio-central.com

ps - also, in browsing through audio-phile magazines, they all seem to be in agreement that vinyl still sounds better than super audio cd and dvd-audio. though i do understand that as deejays we don't use the right needles. though i'm not sure why that's the case.

One last thing.. Indeed the shown triangle wave will be better represented with higher samplerates, but the better representation will not make a difference listening wise.

a'right, check it, it seems that since vinyl records are made from digital masters, that it must be in the extra third party mastering process that makes them sound better, and in theory you should be able to make any digital file sound like vinyl that comes from digital, though maybe the vinyl cutting process adds extra naturally occuring vibrations. but really, since samplers are all at 24/96, recording devices are all at 24/96, and pretty much everything in the recording process is at 24/96, serato scratch live should also be at 24/96, rather than assume it is only used at the end of the line. there are people who would like to manipulate things as though it were vinyl, put the result back into the recording, and then apply further processing to it. and definitely i have done blind comparisons, and also had other people do blind comparisons, and it is clear that either higher sample rates, or higher bit rates, or the combination of the two, do make a difference in the sound. the question is whether it is worth it since most people don't listen to music closely enough to tell the difference, and it seems that a majority of people are even willing to put up with mp3s, and so the difference between cd and dvd-audio is not even an issue worth considering, for them. i was just trying to put in a vote for it, and hoping there would be other people who used scratch live that agreed.

but also, there is no such thing as a square wave as an original audio source (sorry if i misunderstood what was being said). and it is true that lower sample rates, after being sent out by the digital to analog converter, would actually be the smoothest, but the real sound waves in real life are sine waves multiplied by sine waves, multiplied by sine waves, so from a distance the wave might look smooth, but the closer and closer you get, the more and more sine waves you see within the sine waves. these are the sine waves that are missing with low sample rates, and less naturally represented with lower bit rates, and these waves set certain hair follicles in perpetual motion, creating different audio experiences than if they weren't there.

on a third note, i have read two differrent studies that concluded mp3s cause ringing in the ears and that at any given volume, they are more prone to cause hearing damage. that alone is enough for me to avoid all clubs that a dj is djing with mp3s.

overall, things are pretty simple, but it often times seems that people like to create little rules and stick by them when fundementally they are flawed. (i'm not talking about on here but in the blog links) . the whole concept of audio waves and sampling seems like it can be summed up by anybody who took just very basic calculus. if the converters are using specific data points as their original source, there is no way they can accurately recreate the original sampled audio correctly unless they came up with a formula(s) to represent the complete song's original wave which i am 100% positive they are not. my theory is that by putting certain hair sensors in your ears into drastic perpetual motion, while leaving other ones that would normally also be put into fast smaller motions, completely still, causes a severe discrepency in what your brain is expecting, and therefore causes you to tense up your ears and suffer miserably from the audio. this constant tensing up of your ears causes damage to the delicate structure of your ears.

time will tell on how this whole digital revolution affects people's hearing.

also, you can scribble a wave on a piece of paper and not be able to come up with a mathematical formula to represent that wave because no formula exists because it is not a naturally occuring wave. i beleive that our brains and ears are not set up to comfortable hear equivalent sound waves because we evolved in a world of naturally occuring sounds. these "fictitional sounds" cause your ears to tense up, and it is agreed by just about all ents that having your ears constantly tensed up causes damage to the structure of your ear.

there are definitely some discrepencies to the argument, though, since i haven't really thought it out. i still can't figure out why recording vinyl at 24/96 sounds different than recording it at 16/44.1. though needles do pick up frequencies as high as 25hz. or maybe that even though the needles aren't accurately duplicating the vinyl at high frequencies, they are still able to move at high frequencies.

it may just be the bit rate.

Eh, what about the last step? Where do you get 24/96 files from?


You almost got it, a rect. waveform (as any other waveform except the basic ones (sin/cos)) is made of sin waves. So concerning the rect. waveform the more sample-freq you add, the more sine wave will you catch, obviously. But the point is, you cant hear those additional sinewaves! Any sinewave you can hear, would have been correctly represented with 44.1kHz (resp. the half freq), any sinewaves above, which can only be represented by higher sample freq., wont make a difference cause you cant hear them.

I would like to read those studies about mp3s causing more damage than the original audio files.


Oh oh... you are making the same mistake as the guy in the thread, who talks about better reprentation, which is only partly true.

i wish i had the studies, i read the mp3 studies in magazines at barnes and noble, (they were audio magazines) i read one about 6 months ago, and the other maybe 10 months ago. and the study done with the high frequencies, i'll try to find it again. what they did was map peoples brain images while letting them hear ultra high naturally accuring frequencies they recorded from the amazon with special equipment, and in fact, though they could not hear the high frequencies, they could see the brain images change as a result.

here is the link to the high frequency study:
jn.physiology.org

even if you were to believe in the double sample rate theaorem, the fact that needles go up to 25hz, and headphones go up to 30hz, would alone warrant for higher sample rates.

everybody records at 24/96, so that's who you get the files from. just contact the record labels, they'll either give you the masters themselves, but often times thay put you in contact with the producers of the songs. especially with the internet, they don't even need to mail them to you. you can get the songs before they get pressed up on vinyl, and have them in case they don't ever end up getting pressed up on vinyl. then they usually holler back at you as stuff comes out. it's just that i don't have a 24/96 system (i never mentioned that) for deejaying with digital files. but i look at it as though it's still a benefit cuz that's one less sample rate conversion.


i would definitely be interested in what the same mistake i am making is. what part is true, and what part isn't?

check out this link, scroll down a little ways, and look at the three wave form pictures.

studio-central.com

to be honest, i can't honestly tell you why 24/96 sounds better to me and a lot of other people. i guess you have to leave it as a possibility that it is in everyone's imagination. i thought people just wouldn't care, i didn't know people would actually be against it. but i just posted this thing cuz i noticed nobody else had, to try to encourage rane to do it. i know they will eventually get around to doing it, but i was hoping that if enough people were in agreement, that they would do it sooner rather than later. obviously that seems to not be the case. so i will just live with what exists. and stay away from digital files as much as possible for now. luckily i have yet to notice any decrease in vinyl production, though i do sometimes wish i could incorporate more underground and independent artists into my thing. i had the old version of final scratch, which was a piece of junk, so that may have caused me to be more hesitant in jumping to digital for recording. (i'm not against them for trying to make it happen, it just didn't work). my assumption is that rane would rather have something that works really well at 16/44.1, rather than move to 24/96 prematurely, before computers were generally fast enough. in fact, i have no problem using scratch live, it's just since this forum was here, i thought i'd mention the two feature suggestions i'd like to see. i wasn't really thinking about how difficult it would be to achieve. notice i would still use scratch live at 16/44.1 over final scratch at 24/96. the other suggestion i realized later wasn't even for rane to do, but for a turntable company to do.

blogging is not really my thing, and i have clearly gotten a little over my head with this at the cost of letting other things slide. though the info was interesting. though i would still be interested in how the math doesn't add up. my assumption was that digital converters took the data points, and then drew in a smooth line to connect them all, and then output that sound. i understand that that could be completely wrong, and if anybody knows how they actually do it, that would definitely be of interest to me.

Where? All consumer needles I know only go up to around 20kHz. And even then you would need a phono pre-amp which goes til 25kHz. And what about the source? There is a freq limit from the press plant cause it harms the "pressing needle".


Hehe. Nice shot. But open your eyes and you will be shocked what the reality has for you. ;-) Record labels fear piracy, and you expect them to give out 24/96 audio files?


Dont get me wrong, I am not against it if there is some real advantages we get, and right now I dont see any.


The old FS1 sound was really bad compared to FS2 or SSL.

Concerning your 2nd link, that here taken from your link, reflects exactly your misunderstanding (and also my prior misunderstanding):


(And sorry about your 1st link, thats a really huge paper and it will take some time to read+understand it. Results were, yes there are freq. which can not be "heard" cogntivily but the brain shows some kind of reaction. Thats like with very low freq, you cant hear them with the ears, but you feel the bumping bass with your body)


Its a very interesting discussion, even if I also have other things to do in RL ;)

Vinyl sounds better because of the harmonics created by the medium (vinyl).

Yes, I think an accurate way to think of vinyl is that the whole cutting/pressing/playback with it functions as an effects processor.

I think you would only benefit if you were playing your own tunes that you made. Sorry if this has been mentioned but this thread is looonnng.

... and interesting. Definitly worth a read :)

iv'e been through all of this before many times, with you infact! and this is the conclusion i came too. when i was deciding what to get ssl or fs2 this was one of the deciding factors. after researching and testing i discoverd that it made no difference to me. i record my vinyl or rip from cd's that are at 16/44 anyway. also rane is renowned for there extremely high sound quality equipment yet the new ttm57 still uses 16/44. rane know what there doing! imo 24/96 is a marketing tool in playback systems. making your own tunes is a different story though.

You've been through this with me? Are you sure? Cause some of the results I got in this thread was new to me.

not to this degree, you guys are talking some techness. ill read the thread though. ill try to understand :)

Yes. SL should have 24/96 Output Sound Quality & 24/96 Input Sound Quality & stupendously good S/N ratio & ridiculously low distortion characteristics & the analogue valve outputs should be bypassable & the whole unit should be bypassable so that your turntables will pass signal thru when the box is off ......
& all this should have happened 10 years ago.

Use your hears and don't believe all that marketing hype.

nik39&majorp

Bottomline is this:

Most people (here) use mp3 files taken from 16bit/44.1KHz audio sources (CDs). Some will use 24bit/48KHz and very few 24bit/96KHz

So for the vast majority of users a good 16bit 44.1KHz inteface will be more then enough. Most modern day interfaces have a 24bit/48KHz resolution as low-end so 9/10 time your good to go..

For our purpose you do not need anything higher and it will only add to the cost of the interface..

to be fair, 24 bit D/A would eliminate any worries about clipping.
Higher sampling rates are often used during production because many plug-ins work better that way (i.e. EQ).

24 bit helps with clipping if the 16 bit signal isnt scaled to 24 bits

yes.

acme:

true.. I could see teh advantage of 24bit, just not of bitrates >44k1 (with maybe the exception of 48 KHz when you start doing video..)

I don't see advantages with higher sampling rates and SSL - just saying that 96khz can sometimes make sense in the studio.

PLEASE, PLEASE, PLEASE, put 96/24 Firewire in your next version.

The most of proffesional DJs do note the difference in our headphones, monitors, and clubs with big PAs.

Check this post from François K on the way he handles his digital - it is really highlighting:

www.sonicfoundation.net

hey, it's just Francois K opinion.
What about Richie Hawtin playing with SSL at every gig?
Do you think he would compromise his sound?

Having known both men personally, Francois is way more detailed and fanatical about sound quality than Rich.

Maybe. But fanatism is not always good.

For me, a noisy club full of people is not the proper enviromnment to judge equipment.

This is the link to the original discussion with Francois K
www.wavemusic.com

He says that he prefers recording at 24bit/96khz to catch the smooth sound of vinyl - not that playing with 24/96 capable equipment will change anything if your files are standard 16/44.
He even talks about digital "broken stairstepped waveforms", which we know is just a misconception of how a D/A converter works.

I have infinite respect for FK (both his DJ-ing and production career over the last 30 years)... if you're not familiar with his background, listen to some of his productions/remixes from the 80's and note how good his work sounds compared to most other stuff from the period. Depeche Mode's "Personal Jesus" and Terence Trent D'Arby "Wishing Well" (his 12-inch mixes) are two good examples to start with.

I also believe the club and DJ scene in 2006 is sadly backwards in terms of sound quality...maybe it has something to do with the backwards half-assed nature of most of the music being played in clubs nowadays (who really cares if 50 Cent, Cassie or Justin Timberlake is reproduced to a crowd of chodes in high fidelity or not).

On the subject of 96kHz and "stairstepping", however, I believe Francois K is mistaken. The dynamic range and frequency range of vinyl records are both *way* narrower than what 16-bit/44.1 Khz digital is capable of. It's standard practice in vinyl mastering to filter the highs above 18kHz or so, and if you find a record with 50-60dB of dynamic range you have a *very* good pressing (compared to the 96dB old-fashioned 16-bit digital provides). The whole "digital stairstepping waveforms" myth was debunked long before the first CD players hit the market, and is easy for anyone to verify using an inexpensive oscilliscope.

The only things you might be better-reproducing with higher sample-rates (for recording from vinyl records) are small ultrasonic ticks from dust or vinyl imperfections. Personally, those noises give me no enjoyment, and hardly seem worth more-than-doubling file sizes and processor load.

The point being missed here is that many DA converters in soundcards etc are fairly cheap in quality. Yes even studio grade cards apart for Digidesign HD interfaces and that kind of priced item (Lucid etc). 24 bit sounds better because your DA's are able to capture more information. In saying this though many cards are not able to capture sign waves well at the cards full resolution starting from 96k @ 24 bit and up.A happy medium of 48K @ 24 bit will work well on most cheap cards ($500 - $1500). If you use a decent DA it will sound way better at a lower bit rate and sampling rate that a cheap card at a high resolution.

Without preprocessing a 16bit audio file is not going to sound any different on a 24bit interface just because of it being 24 bit.

Allow me to bump this up. I'm not normally one to cross-post, but my thread on whitelabel.net hits home with a similar issue

beta.whitelabel.net

slick

I agree with this, this is the ONE thing that will make me rush out and buy Serato..... and higher if it is possible...

I dont understand the theory, but i do know for the past 2 years I have been recording DJ mixes off vinyl at 44.1/16 and have been through so many mastering presets etc trying to get the sound as clear, with as much "space" around the instruments as I hear on vinyl......

The other day I upped the sample/bit, and was ASTOUNDED at the difference... there is NO DOUBT to me now, higher sample and bit rate improve the sound. As I say I have no theory.. its just the difference after listening for 2 years to a lower sample rate was immediately noticeable..cymbals no longer pierced, bass was more defined - I was locked into the headphones in astonishment.

THe latest Mixmag has the makers of FUNKTION 1 speakers slating DJs for using mp3s, plus a D & B crew who has banned mp3s and CDs on their nights, stating they sound bad..

THe mp3/CD backlash is starting - DJs need to be able to step up and provide the high quality sound that new audio standards, and new computer advances allow...

At the moment there is no point me recording my vinyl at high rates only to put it back out at CD only quality through serato. CD quality is the old standard now, and not especially high - there is SACD now....

THere is a new audio standard on the computers now also....and computers can handle higher and higher processing demands, and competitors bringing stuff out that can handle 192/24 this needs to be addressed soon...Looking at serato software there is nothing that can beat it... Im holding out for the new box, its got to come one day....

do that somewhere with an audio outputs QUALITY!!!! (louder sound, better converters)
and you can leave 16/44.1
we need PRO quality 2 play records at bigroom dancefloors and quality club sounds!!!!

if they update the SL2, you'd have to have 24bit 48Khz to compete with TSP

Now pro djs migrate to TSP, because TSP outputs quality better than SSL1.
SSL2 must be more professional sound interface like ECHO or RME!!!!
(i mean sound brighter and louder)

I'd say the Native instruments osuncard would be better than the echo, probably not so the RME (RME has the best Asio drivers)

Right on I use the RME fireface 400 for DJing. Originaly bought it sounds os good that I've been rocking events with it in the past few weeks.

Frankly the SSL box problem is not it's bite rate/frequency, what needs to be i,nproved is the quality of the hardware components. An mp3 encoded in 44.1 khz won't sound any better played back at 96 khz... These high bitrates only improve time code precision and playback of high quality encioded tracks (usually studio masters and such)

That's wrong, since if you are using a control signal you are limited by the quality of the control signal itself, which is for vinyl limited by the styles etc., for CD limited by the CD specs itself.

The higher bitrate won't help with the timecode but if Serato starts adding effect in side the program when you run say an aiif through the effects, you'll will see less degradation of the sound. MP3s should be banned for Scratchlive... there I said it

Good stuff, I hit the report button for ya'

;)

damn you Nik39

which program you are use for djing on RME fireface 400?
pc or mac?

That's wrong, since if you are using a control signal you are limited by the quality of the control signal itself, which is for vinyl limited by the styles etc., for CD limited by the CD specs itself.

Higher bitrate means more frequent timecode signal and improves the real time feel. Specially with fast scratching on both an Edirol FA-101 and now the RME fireface 400.



MixVibes CROSS vinyl on a Mac, it's just betatesting I don't use it out yet. I'm sticking with MixVibes PRODUCER 7 (Win XPBootcamp) for clubs with vinyl, I've also been using Traktor 3 with a VCI-100 in clubs that have no decks.

Not here to do the MixVibes vs Serato thing, but I'm just saying sound quality should be a major concern for all DJs and the SSL box is not a top choice in the ASIO sound device market...

Any tips on SSL coming up with a new interface ? Maybe firewire or USB2 to handle high quality signals and multiple decks ??

Maybe you would like to define the term "higher bitrate" first.

you are right nik39 i meant higher sampling/output frequency. e.g. switching from 44.1 khz to 96khz

I don't know if changing the bitrate (from 16 to 24 bits for example) can improve timecode decoding.

Frequency... is limited by the needles.



Limited by NSR - Vinyl is about (IIRC) 60dB, 16Bit offer 96dB. So again vinyl is the limiting factor. Even if you use CD players the control signal medium is limited.

I see a lot of confusion here between ADC and DAC samplerate and bitrate.

While I see no point in improving the input stage (except for sampling maybe), the output stage could probably be better. I'm not saying it's bad!

A naive question: can a 44.1 kHz 16 bit file be rendered with improved quality through a sound card with greater samplerate and bitrate? Or is the competition ekwipt is talking about just a war of numbers?

I'm surprised no one has hacked/modded a SL1 yet, e.g. to add a better digital clock or change the capacitors or whatever... I've googled and couldn't find anything :-(

Whot?

Yeah, it's not uncommon to have the clock replaced.
Example: there are 2 clock kits available for the Behringer DCX2496
www.dcx2496.fr
This site also lists a few other tweaks (replace the Cirrus Logic CS8420 by the CS8416, filtered power supply etc.)

Sorry, the site is in French :-/

You can also find clock kits, like this one:
www.hagtech.com

I hope this will give some inspiration to people out there :)

I know zero about stuff like this...but could I buy that clock kit and get someone to mod a SL1 box for me?

I think its a little bit silly (but understandable) to rip vinyl to higher rates in order to try and capture the "vinyl sound". It would be the same as using higher rates to capture that cassette tape sound. The only difference is cassette sucks while vinyl has a pleasing sound. It makes a whole more sense to try and digitally (or otherwise) recreate that pleasing vinyl sound. Even starting with 16/44.1 you have more range then vinyl. So if you could process the 'vinyl sound' in, then over all you would have more control and a better sound.

We are never going to get any records at better then CD from any labels ever. Not until a better format overtakes CD (if that ever happens). So I just cant see the need for SSL to offer 24/96 out, whatever the cost.

extremely little benefits

I would much rather see digital emulation of the 'vinyl sound'

+1, +1.

but that's impossible, vinyl is analog... why not get a higher bit and Khz for scratchlive it will help in processing for example keylock and effect in the future...

SL1 won't go past Scratchlive 2.0, there's no need for it, it's pretty much used all the feature set is capable of up until now, loop roll feature will be implemented and then maybe bugs can be ironed out, possibley add album art and tie cues to loop points, better midi... that's it

I'm looking forward o the Serato/Ableton partnership

Do you know this to be the case or is this a hunch?

I actually think you're probably right...I'm just curious if you have any inside information.

hmm, I think we will be surprised at how powerful the SL1 actually is.



Its impossible to digitally emulate analog sounds? really?

I think processing effects internally is somewhat of a dif subject then the output. How much better would the effects sound? I dont know. Would it really make that much dif if they where processed with hi rates and then sampled down to 16/44.1 for the output, if that even makes sense? I dont even think we should put effect before the box, but if we do, and quality is an issue, I think it would far more efficient to use an external effects box in the first place.

I wish cd was better quality from the start. But I have a hunch we will live to see a format change with higher rates. Format changes make lots of money.

There's a lots of really good analog emulation in the digital recording world...makes sense that vinyl emulation would be possible in the dvs world.

Just a hunch. If Serato simply updated the soundcard for the SL1 and let Scratch Live stay as 2 decks than I'm sure the product would have another few years of life left in it....

The only things that need major improvements is the internal part in Serato, if it was updated for full control with midi controllers it would be a perfect package for hip hop DJs (it is now).

The SL2 could be designed so much smaller I'm sure

SL2 - SL1 Serato Scratch Live

SL4 - Serato Live (Ableton) Unlimited control of tracks "Dependent on computer used of course"

That's what I'd do anyway

Just got an email from the Black Lion audio guys:

You might want to look into the op-amps, converter decoupling, and capacitors in the analog stages. Unfortunately we are not taking on custom modifications at this time.