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is 24/96 worth the sacrifice in speed/space?

Discussion in 'Recording' started by took-the-red-pill, Jun 28, 2005.

  1. took-the-red-pill

    took-the-red-pill Active Member

    So I did an A/B through my cans of 24/96 and then 16/44.1. The difference made me wish we weren't still using that 20 year old standard on our CD players. Yoiks! I had no idea.

    The question is, do we actually get a better recording by inputting it at 24/96 and then dithering down, or are we fooling ourselves and we should just record it on 16/44.1.

    Not trying to start a war, just asking.

    Thanks
    Keith
     
  2. David French

    David French Well-Known Member

    Covered lots of times. Try searching for threads.
     
  3. Markd102

    Markd102 Well-Known Member

    Well I'm still stuck with the good old Digi001, so 48k is my maximum.... but two friends who run big Digi HD3 systems still only record at 24/48 for bands, but one had a string quartet in last year and he used 192khz for that.
     
  4. slicerecords

    slicerecords Guest

    Superb sound quality doesn't come from what sampling rate/bit depth your working at--it is determined by the quality of the instrumentalist, then the instrument, then the room, then the engineer, then the mic, then the preamp, then the conversion quality (not determined by what sampling rate your at). 44.1K/24-bit is as high as I'll go if I know the final product will be a CD- Of course that changes if your doing A/V work, or DVD-A work. Dithering is at a place where working at 24 then going to 16 is not a bad thing---but SRC is not (at least to me) beneficial to the final project--if anything going from 48k to 44.1 hurts a project more than recording at 44.1 and staying there. Of course there are millions of opinions on this topc.
     
  5. David French

    David French Well-Known Member

    Here is the somewhat 'end-all' thread on the subject.

    http://recforums.prosoundweb.com/index.php/t/288/0
     
  6. Fooldog01

    Fooldog01 Guest

    Unfortunately I dont think the link really answered the question. I have been wondering this myself for quite sometime, but I havent found anything that definitively gives me any answer. 96khz takes up more space than 48, and 48 takes up more space than 44.1. Now I dont know about the difference in processing, but I think it has been settled that 24bit dithered to 16 sounds better because of a number of reasons... however, does 24/96 dithered down, sound better than 24/48 dithered down, or 24/44.1? Is there any benefit to using 96 instead of 48, or using 48 instead of 44.1?

    Its not about the signal chain... its a specific question designed to clear up a particular dilemma. Hopefully this clears things up a little and maybe we can draw some conclusions.

    Sweet
     
  7. Cucco

    Cucco Distinguished Member

    You know - Dan makes a valid point (or alludes to it at least) that there's the science involved and then there's the pure live testing (listening).

    Many people claim that there is no information to be had above 20kHz - many people claim otherwise.

    The interesting fact is, in blind tests, people have been able to identify the same program material recorded at different sample rates and identify whether they were 44.1 or high sample rates. Obviously, something is going on and whether it's the distortions that Dan references or some actual program material, it is discernable.

    Of course, the converse is also true - some people actually preferred the sound of 44.1 in numerous of these listening tests - so - what to do???

    Personally, the majority of the projects I record and mix are in 24bit / 44.1 kHz. If I do a major orchestral project, I will bring up the sampling rate to 88.2 or 96k - sometimes 192 or DSD.

    Ultimately, is there a difference? Sure, it takes up more of my harddrive space and I can hear a subtle difference. But, I can't say that it's such an improvement that it makes me die to use higher sampling rates. Now, I'm a firm believer in DSD however. I know a lot of people who crap all over DSD because of its theoretical limitations, but frankly, it just sounds better and there are plenty of scientific reasons why. (What I find to be funny is that the opponents of DSD are often the proponents of 24/44.1 and their argument for 44.1 is that there is nothing to be had outside of 20k. Then they turn around in the very next breath and say that one of the serious drawbacks to DSD is the excess of "out-of-band" noise centered above 60 kHz. Wow, now that's a contradiction in philosophy!)

    J.
     
  8. CoyoteTrax

    CoyoteTrax Well-Known Member

    It's weird, I've heard "definite" differences when notching out 22kHz by only a half dB on a final mix. I didn't think I could hear 22kHz but was proven wrong.

    Ya gotta hear it to believe it.
     
  9. JBsound

    JBsound Guest

    24/96 on good converters sounds great, but I end up going 24/44.1 about 95% of the time. It just saves so much disk space and processing power. I also feel like some of the mediocre converters do not necessarily sound better at 96k than they do at 44.1.

    If I'm doing only a few tracks and it's classical or acoustic music and I have some decent converter's then I go 96k...but most of the time on my digi002r I will just go 44.1.
     
  10. Nika

    Nika Guest

    J,

    We meet again. I hope you're well. To clarify the perspective, the issue with the out of band noise in DSD is three fold:

    1. many proponents of DSD tout its high frequency response. I don't bring up the out of band response of DSD in order to say that I believe we can hear it, but rather to discount that posit on their behalf that that is why it is better.

    2. the out of band noise at high levels contributes to distortion in amps and speakers that are ill-designed to handle that kind of signal. Tweeters become particularly non-linear in that range, and the non-linear behavior causes distortion in our audible frequency range.

    3. in order to prevent this excess out of band noise Sony recommends putting a low pass filter on. This filter, however, does not get the benefits of a digital, FIR, linear-phase filter as it needs to be in the analog world. Instead, this filter causes phase shift of audible frequencies as it inherently must.

    Yes, the out of band noise is an issue and continues to be a source of concern regarding DSD, but not because we speculate it can be directly heard.

    Nika
     
  11. took-the-red-pill

    took-the-red-pill Active Member

    Wow! I thought this was a simple question. Turns out it's a topic of much discussion and many opinions.

    I guess I'll do some recordings at 24/96, 24/44.1, and 16/44.1 and then burn some CD's and let my ears decide.

    If this thread isn't dead yet I'll report my findings here, at least as far as my ears and equipment can tell anyway.

    Thanks guys
    Keith
     
  12. Midlandmorgan

    Midlandmorgan Active Member

    Simple question? I've seen flame wars, fist fights, couple of stabbings, and at least one shooting over this very topic.... :wink:

    Seriously though, the general consensus that I've been able to gather is that 5 years ago most people HAD to have 96K or 192K capabilities...and those exact same people are now reporting they stick to 24bit/44.1K as the cost/benefit ratio is drive space is just too high...even with the higher rates, there are those who argue that the better conversion systems running at 44.1 exponentially sound better than lesser conversion at 96K....

    I tend to use 96K ONLY when recording string ensembles or acoustic jazz...for the typical rock/country production, its 24/44.1 all the way....for radio ads, PSA voiceovers, etc, I don't even think twice - its the lowest I can get away with and still provide a good sounding mix....

    Really long story short: you are correct. Let your own ears decide...IF you think the higher rates are worth the investment in drive space/lowered processing abilities, then go for it.
     
  13. Cucco

    Cucco Distinguished Member

    Yeah, in truth, I would never refer to this as a simple question.

    If you plan on doing comparisons, make sure that you are taking the signal from the exact same source. I have a setup which has allowed me to do just that.

    The RAMSA WZ-AD96M uses ADAT Lightpipe and AES outputs. With 2 of those hooked up to my True Systems Precision 8 (which has 2 identical and active sets of balanced output) has allowed me to perform these comparisons with the identical signal present at both ADs and thus at mix down. In my experience, I've heard some differences, but bear in mind, all of the differences have been subtle and noticeable only on my high-end system or monitoring set-up, not with standard playback equipment.

    Nika-
    How do you do? All settled in to the new life yet?

    For anyone who's interested, Nika and I have pretty vastly differing opinions on some of the finer points regarding high sample rates and specifically DSD. The difference is, he's done a bit more scientific research into it than I have. It's an interesting read to say the least if you can find the threads.

    You make some good points here, but I know you are aware that, with the proper playback system, many of these limitations cease to exist. True, many tweeters do become quite unpredictable at these higher frequencies - a problem solved one of two ways: a simple analog filter analogous to a crossover with no hand-off to a higher frequency band or specific materials used for their more linear production of UHFs. Of course, the Bose paper cone tweeters aren't good at this, but several of the aluminum, titanium, ceramic/metal hybrids do a fantastic job of resisting distortion and break up well into the octaves above theoretical audible limit.

    Many amplifiers are now being designed and built to handle this kind of bandwidth as well. Of course, all of this suggests that one has taken the time, effort, and money to piece together a system capable of these feats.

    Of course, much of the debate re: DSD (quasi-PSM) vs PCM are akin to those debates of higher vs lower sample rates. It really boils down to the single bit modulation and the theoretical advantages and problems inherit in it. Experts have come out on both sides of the fence on this one with facts to prove that DSD both sucks and reigns supreme. Personally, due to the conflict and my lack of all the information, I rely entirely on my ears. Thus, I enjoy DSD and SACDs.

    So, I guess the moral of the story is - get all the facts you can (which should be easy, there are several good books on the matter) and avoid all the hype/opinion on the matter. Then, you be the judge - you should be the toughest person to convince. And ultimately, if you ever lose money on a gig b/c a client demands 24/192 - then RUN, don't walk to your nearest store - drop $3000 on the newest converter and regain your lost business.

    J.
     
  14. There's proof you're from Texas.... we've had that problem in Oklahoma too...

    Anyway, I've probably already made it clear that I don't record digitally, but that's besides the points I am about to attempt to make based on my knowledge. First of all, it is a standard procedure to broadcast time signals via radio at 60 kHz. These signals are typically used to update the date and time automatically on many electronic devices (cell phones, for example, usually have a "feature" to automatically update the time itself). This can be a possible explaination the out-of-band noise mentioned in an earlier post. A time code is synchronized with the 60 kHz carrier and is broadcast continuously at a rate of 1 bit per second using pulse width modulation. The carrier power is reduced and restored to produce the time code bits. The carrier power is reduced 10 dB at the start of each second, so that the leading edge of every negative going pulse is on time. Full power is restored 0.2 s later for a binary “0”, 0.5 s later for a binary “1”, or 0.8 s later to convey a position marker. The binary coded decimal (BCD) format is used so that binary digits are combined to represent decimal numbers. The time code contains the year, day of year, hour, minute, second, and flags that indicate the status of Daylight Saving Time, leap years, and leap seconds.

    Also, the changes that can be "heard" by adjusting, say, 22 kHz aren't necessarily heard, but rather perceived much like subharmonic frequencies.

    These are just things that I learned about when I should've been trying to pass high school. It's just a hunch, though, considering that I nearly dropped out of high school so I can't possibly know what I am talking about. (Sorry I'm venting there, I just had a rough night at work about how my supervisor is convinced that hydraulic equipment doesn't need a fluid and filter change after over 5 years....he mentioned that I couldn't know what I was talking about and how he had a degree....) Anyway, the following link should help diffuse a little information about interference. http://

    Notice how much falls between 9 kHz and ~192 kHz.

    Anyway....
     
  15. took-the-red-pill

    took-the-red-pill Active Member

    Cool, 8)

    Seems to me the best way to test this specific sample rate issue is to record the same signal in 24/96(or similar) and 16/44.1. Cut them to a regular CD-A and then A/B the results on a system like what Joe Sixpack might have in his house, or his car.

    Isn't that 'real life' situation the one that matters most? I hate to be a poop, but who cares if we can hear the difference through our squeaky clean, high end signal chains, since ultimately that isn't a 'real life' situation.

    It only really matters if Joe's less than perfect D/A converters, average monitors, running a 16/44.1 CD-A can spit out the difference in such a way that it's discernable to the naked ear. All else appears, to me at least, to be pretty much academic.

    Anyway, many thanks for the replies. I'll add that 24/44.1 to the list of things to test out, as some of you have suggested.

    By the way, as downloading explodes, and music cheapens, I think the importance of purity of signal is going to get pushed further and further down the list. This whole discussion starts to look a lot more like intellectual masturbation in that light.

    ...but that's a topic for another day...

    Keith
     
  16. Arrowfan

    Arrowfan Guest

    If you plan to be using a lot of plugins, then definitely stay at 48Khz or you'll run out of CPU/RAM quickly.

    If you're recording more than 8 tracks at once its probably best to record at 48Khz (unless you have an FW800 bus supporting audio interface like RME's Fireface800 which can handle the bandwidth).

    But going from 48Khz to 96Khz should increase high end clarity and tighten bass somewhat, so whenever possible use it.

    Disk space really shouldn't be an issue, if it is get some external firewire hard disks. Lacie/Porsche makes some great ones, but make sure it doesn't have any USB ports - just 2 FW ports so you can daisy chain them. They've got a 160GB, 7200 RPM, 2 FW ports for less than $130.

    When you archive sessions made at 96khz, definitely remove all the out-takes and un-used audio (SX has a nice command for this "Save Project To New Folder").
     
  17. Cucco

    Cucco Distinguished Member

    Arrowfan -

    You've stated some of the untrue myths about higher sample rates --

    High SRs will not improve bass frequencies. For awesome bass, you could use a sampling rate of 400 Hz and get the EXACT same bass (from 0 Hz up to 199 Hz) as you would if you used 384 kHz sampling rate. Remember, you only need 2 points on the graph to identify a frequency. No additional points will help - they will only help identify higher frequencies.

    As well, it takes no significant additional processing/RAM power to process 96 vs 48 vs 44.1.

    Keith -

    Ultimately, yes it is advisable to test anyway that you can. However, NOT testing it on the highest end system that is available is a mistake. I assume that every disc I make will be played back through:
    The finest CD transport
    Awesome D/A converters
    Krell or Mark Levinson Amplifiers
    B&W or Revel Loudspeakers
    over $10,000 cables

    Why - some of my clients (and probably yours too) have this kind of set up. If you want them (the people closest to being "in the know") to think that your disc sounds like $*^t b/c all you did was make it sound good on your Ford's stereo system, then by all means - ignore this demographic.

    Moral of the story - never skimp on the monitoring chain - EVER. :D

    J.
     
  18. JoeH

    JoeH Well-Known Member

    Seems to me the best way to test this specific sample rate issue is to record the same signal in 24/96(or similar) and 16/44.1. Cut them to a regular CD-A and then A/B the results on a system like what Joe Sixpack might have in his house, or his car.

    Well, that's a start, but only part of the process. You also open up the "can of worms" that includes sample rate conversion (which one is best?) and dithering down from 24 to 16 bit. (I know what "I" like, but....)

    All of these steps "corrupt" the scientific method in determining which one is "best". By the time it's all boiled down to Redbook CDs, who knows what's happened to the sound, and where/how/why?

    In a perfect world (not the way people listen, anyway), you could go with a DVD-A or DVD-ROM or HD wav file playback to first find out if there's an appreciable difference for you at the top of the listening chain. (let someone other than the listener keep track of which track is which.) To be scientific about it, you'd have to have recorded all different versions at the same time or under strictly identical conditions, with similar gear - 16/44, 24/44, 16/96, 24/96, etc.

    Keep your results noted there, and then continue with the CD making process, noting carefully what steps were taken.

    Make your CDs with the SRC and dithering of you choice, compare the same tracks again (also with the "control" person aware of which one's which) and see which one you like, what the differences are, etc. No one ever REALLY takes those steps when making divine "pronouncements" and comparisons, and IMHO, by the time anyone's done any real listening, it's way too far down the chain to be scientific enough to matter.

    I like to record at the highest level practical for the project at hand, take the best care of it throughout the DSP and mixing, and then master it with all the TLC possible down to 16/44. (Always keeping the hi-res master on file for "someday"....whatever that might bring.)
     
  19. took-the-red-pill

    took-the-red-pill Active Member

    Jeremy, point taken.

    Actually I don't necessarily agree. I don't know anyone in my world who possesses a stereo system even remotely approaching what you are describing. I know guys with money, and I know guys who really care how music sounds, and they are still listening through the stuff you can get at Future Shop for a few grand. So to create music to the standard you describe would not be a 'real world' situation for me.

    It would be kind of like the government saying "We need to assume when we build all our roads that a formula one car can go screaming down it at 200MPH and not hit a hole or bump."

    That would be nice, but hardly practical, since in reality that road will never see a formula one car, so they let the odd bump and pothole go because Joe and Jane Sixpack can drive it in their Civic and it will get them where they're going.

    I'm not advocating a weak signal chain on recording, mixing and mastering, merely stating that we need to consider the average stereo's capacity instead of overkilling, or adding redundancies. I'll just create recordings of the highest quality I reasonably can and if there are 3 people out there who are offended, because they have 25K in listening gear and my CD has holes in it due to my D/A converters or sample rate...well I guess I'll include that in my refund policy. Currently I don't have a 'list of clients' who have expectations, so I have more room on that one than some.

    Well anyway, this is beginning to wander off topic, and I did say I wasn't trying to start a war, so I'll shut up until I've done some testing.

    Cheers
    Keith
     
  20. Cucco

    Cucco Distinguished Member

    Hey Keith -

    I don't think you're starting a war or straying off topic - don't worry.

    My logic behind aiming for the highest quality is simple -
    True, you may not have clients with that esoteric stuff now, but you might and hopefully will. You don't want to alienate them because you're designing for the lo-fi or mid-fi crowd. If you provide the absolute best sound possible, everyone will be satisfied from the dude with the RadioShack boom box all the way up to $100K invested in audio equipment.

    My point is, don't exclude and of the market segment.

    J. :D
     

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