Here's another reason to record at high sampling rates. However, it is rather contrived and currently doesn't exist in the real world AFAIK.
If you have an accurate numerical model of a nonlinear distortion, and you are able to invert the model, then you can remove the distortion from a signal. The signal will intermodulate with itself in the presence of that distortion, generating ultrasonics. Those ultrasonics are probably going to be necessary in order to run the distortion cleaner accurately. (Or, if you know more than that, perhaps you can autodetect the distortion model based on the ultrasonics and work from there.)
This could apply both to recording (to correct nonlinearities in analog media) or to playback (to correct nonlinearities in speakers). But you'd need scarily accurate simulations.
Full Version: Why 24bit/48kHz/96kHz/
I'm kind of late to this thread, and I haven't read it in its entirety, but generally some intuitive thoughts supportive of high resolution audio formats would be:
1. Headroom: I would like to see recordings with 90dB nominal SNR where there was still 40dB headroom for loud dynamic passages. The nominal recording level should be far, far below 0dBfs for real tangible dynamic headroom, but still be perceptually transparent in the "nominal mode".
2. Why not? It's no problem with modern computer technology.
3. It is always much harder to improve analog performance than digital, in modern submicron processes more so than ever, thus the error contribution from the format itself should preferably be negligible, even when budgeting for something on the limit of human hearing capabilities. Then, the entire error budget could be allocated to the analog parts, instead of being eaten up by the digital encoding and processing.
1. Headroom: I would like to see recordings with 90dB nominal SNR where there was still 40dB headroom for loud dynamic passages. The nominal recording level should be far, far below 0dBfs for real tangible dynamic headroom, but still be perceptually transparent in the "nominal mode".
2. Why not? It's no problem with modern computer technology.
3. It is always much harder to improve analog performance than digital, in modern submicron processes more so than ever, thus the error contribution from the format itself should preferably be negligible, even when budgeting for something on the limit of human hearing capabilities. Then, the entire error budget could be allocated to the analog parts, instead of being eaten up by the digital encoding and processing.
QUOTE(dekkersj @ Jan 19 2007, 11:55) 
Quick update.
I am replying to the original question: "Why 24bit/48kHz/96kHz".
A few weeks ago I visited a member of another forum and as a sort of sanity check I played a sweep at -89 dBFS. Normally it stays silent, meaning that 16 bits is enough. But in his case I could hear a very small fraction of the sweep. That was because of the quite large playback level. We could not talk with each other when listening to his set. Funny thing was that he could not hear a thing of the sweep.
Conclusion: at normal levels, 16 bits is enough.
Regards,
Jacco
That's probably because his already damaged his hearing by listening at such high volume levels.
QUOTE(ilo @ Jan 20 2007, 01:44)
I'm kind of late to this thread, and I haven't read it in its entirety, but generally some intuitive thoughts supportive of high resolution audio formats would be:
1. Headroom: I would like to see recordings with 90dB nominal SNR where there was still 40dB headroom for loud dynamic passages. The nominal recording level should be far, far below 0dBfs for real tangible dynamic headroom, but still be perceptually transparent in the "nominal mode".
Have you heard/got the Hi-Fi news test CD, which includes a single fire cracker / firework being set off, and the crowd's reaction to it? It's a great example of dynamic range. Here is a short extract:
Click to view attachment
That's from a CD, so is "only" 16-bits.
The problem I have reproducing this realistically (i.e. to sound as if I was stood in the crowd) isn't the 16-bit limit, but my amplifier, speakers, and neighbours!
Cheers,
David.
It is possible to create digital content with practically any snr wanted. I could take a regular cd recording at 16 bits, decide that I wanted to pad down the middle section by 96dB, and effectively have a 32bit dynamic range mix. I cant see that this is anything that anyone wants to do. Otherwise, you are practically limited by the microphones, pre-amps and rooms of this world. I dont think that it is possible to capture a live acoustic performance with 32bits of precision at a reasonable samplerate today.
Of course, if you listen to techno or some other purely synthetic music, then all usic could be rendered as a mathematical model inside a vst plugin as 32bit, 64bit or even 128bit floating poit precision numbers. But how important is that in the big picture?
As long as adc boxes capture the signal from microphone/preamps exceeding their specs, I see no place for "headroom increase" as mentioned above. The problem lie in a signal chain of limited SNR (mike, preamp, dynamics processors, amplifiers, rooms etc). And even that chain seems to satisfy our limited hearing senses.
It is interesting with "extreme dynamic" reproduction. But I think that we need better microphone/loudspeaker technology before blaiming the poor CD player. And given the choice, sound engineers hired for their subjective tastenearly always _reduce_ dynamics on live and recorded music, at least partially because that is what sounds better.
-k
Of course, if you listen to techno or some other purely synthetic music, then all usic could be rendered as a mathematical model inside a vst plugin as 32bit, 64bit or even 128bit floating poit precision numbers. But how important is that in the big picture?
As long as adc boxes capture the signal from microphone/preamps exceeding their specs, I see no place for "headroom increase" as mentioned above. The problem lie in a signal chain of limited SNR (mike, preamp, dynamics processors, amplifiers, rooms etc). And even that chain seems to satisfy our limited hearing senses.
It is interesting with "extreme dynamic" reproduction. But I think that we need better microphone/loudspeaker technology before blaiming the poor CD player. And given the choice, sound engineers hired for their subjective tastenearly always _reduce_ dynamics on live and recorded music, at least partially because that is what sounds better.
-k
QUOTE(2Bdecided @ Jan 22 2007, 04:30)
The problem I have reproducing this realistically (i.e. to sound as if I was stood in the crowd) isn't the 16-bit limit, but my amplifier, speakers, and neighbours!
This is a great quote! Every time I read/hear people talking about how 24/96 is necessary for sound quality, I'm constantly reminded of your post
the problem isn't so much the 16bit dynamic limit, rather music being mastered to have 0.5db of dynamic range and headroom. You could have 24bit dynamic limits, if the music is mastered loud w/ no headroom and dynamic range, it will still sound like crap!
QUOTE(William @ Dec 29 2005, 21:45)
Yes, I have searched the forum.
Yes, maybe I am dumb.
But it seems I cannot find the answer.
Why do we need 24bit/48kHz/96kHz/192kHz if 16bit/44.1kHz is good enough? Are there any situations that 16bit/44.1kHz simply cannot satisfy? In other words, is there any real need for the higher bit depth and sampling rate?
Thanks for answering.
Simple: 16bits gives you: 20log (2^16bits) = 96.32dB dynamic range, fine for listening, and mixed materials (mixed CD output).
24bits gives you: 20log (2^24bits) = 144.49dB dynamic range which covers the dynamic range (difference of quietest and loudest points of PGM) of stuff like an orchestra (with up to 120db dyn range), so to capture all nuances, we need high dynamic range.
higher dynamic range means lower noise floor, and when mixing multiple signals, multiple low noise floors combine to give increased noise, so we want to make this as low as possible when mixing.
Higher khZ generally means higher quality - better image representation through shifting the Nyquist frequency higher up (well beyond audible). However, if you have shit-hot (eg apogee) DACs, you don't need super high sample rates (due to their oversampling functions). A great DAC at 44.1 is as good as an average DAC at 192khz. Ask Bob Ludwig about this.
QUOTE(wildnewt @ Feb 21 2007, 14:08)
QUOTE(William @ Dec 29 2005, 21:45)
Yes, I have searched the forum.
Yes, maybe I am dumb.
But it seems I cannot find the answer.
Why do we need 24bit/48kHz/96kHz/192kHz if 16bit/44.1kHz is good enough? Are there any situations that 16bit/44.1kHz simply cannot satisfy? In other words, is there any real need for the higher bit depth and sampling rate?
Thanks for answering.
Simple: 16bits gives you: 20log (2^16bits) = 96.32dB dynamic range, fine for listening, and mixed materials (mixed CD output).
24bits gives you: 20log (2^24bits) = 144.49dB dynamic range which covers the dynamic range (difference of quietest and loudest points of PGM) of stuff like an orchestra (with up to 120db dyn range), so to capture all nuances, we need high dynamic range.
How many systems are actually able to represent that dynamic range, given inevitable toom-noise and the pain-threshold?
QUOTE
higher dynamic range means lower noise floor, and when mixing multiple signals, multiple low noise floors combine to give increased noise, so we want to make this as low as possible when mixing.
So by how many dB do you think that the noise-floor increases if I mix 8 16-bit streams into one 16-bit mix at high internal precision? Assuming that the noise is white uncorrelated, and that signals are uncorrelated.
QUOTE
How many systems are actually able to represent that dynamic range, given inevitable toom-noise and the pain-threshold?
higher dynamic range means lower noise floor, and when mixing multiple signals, multiple low noise floors combine to give increased noise, so we want to make this as low as possible when mixing.
So by how many dB do you think that the noise-floor increases if I mix 8 16-bit streams into one 16-bit mix at high internal precision? Assuming that the noise is white uncorrelated, and that signals are uncorrelated.
don't confuse dynamic range and physical loudness which are both measured in dB. PGM with 144dB dynamic range doesn't necessarily mean the loudest parts of the it will be represented/reproduced at (physical loudness) 144dB.
Without numbers or measuring the 16bit streams, what type of dithering used etc it's hard to say exactly. but in a well designed studio it's possible to hear the noise-floor on one 16bit channel (and a noisefloor of -80dB). never mind when you mix 8. the standard of listening environment required increases dramatically if you want to hear the noise-floor in a 24bit stream.
Unless anyone understands the maths and reasonings behind precision and dB, it's easy to get confused or follow misconceptions.
QUOTE(wildnewt @ Feb 21 2007, 14:02)
Unless anyone understands the maths and reasonings behind precision and dB, it's easy to get confused or follow misconceptions.
Oh, I don't think there's anyone in HA who understands maths
Seriously, at least read some of the thread before jumping in with a half baked answer to a question posed on Dec 29 2005!!!
All your points have been rebutted already. 16-bits is sufficient for delivery to almost any conceivable home listening situation. Mixing can and should take place at higher resolution, and no one has argued against this.
Cheers,
David.
I've been trying to learn about all this stuff from reading and starting at the fundamentals. In this pursuit I read all 17 pages....
Until a couple of posts ago, there was something that was confusing me:
'don't confuse physical loudness with dynamic range...both in db'
Thanks for that!
eeesh...17 pages...that was a hard fought piece of important intel however...now I can continue learning.
Until a couple of posts ago, there was something that was confusing me:
'don't confuse physical loudness with dynamic range...both in db'
Thanks for that!
eeesh...17 pages...that was a hard fought piece of important intel however...now I can continue learning.
QUOTE(w1L50n @ Feb 21 2007, 07:57)
'don't confuse physical loudness with dynamic range...both in db'
Thanks for that!
Thanks for that!
I hope you're kidding!
QUOTE(knutinh @ Feb 21 2007, 08:33)
QUOTE(wildnewt @ Feb 21 2007, 14:08)
Simple: 16bits gives you: 20log (2^16bits) = 96.32dB dynamic range, fine for listening, and mixed materials (mixed CD output).
24bits gives you: 20log (2^24bits) = 144.49dB dynamic range which covers the dynamic range (difference of quietest and loudest points of PGM) of stuff like an orchestra (with up to 120db dyn range), so to capture all nuances, we need high dynamic range.
How many systems are actually able to represent that dynamic range, given inevitable toom-noise and the pain-threshold?
And how many audience members at orchestral concerts actually experience 120 dB of dynamic range? Answer: none. That figure (probably from Alton Everest's book) comes from close-microphone placement.
Heres's from Everest, 4th edition, citing Fielder:
(condition: base SPL/dynamic range/peak SPL )
piano solo 13dB/90dB/103dB
typical classical symphony (tcs), w/audience: 13dB/100 dB/113dB
tcs, w/o audience: 8dB/105dB/113dB
tcs, close mic: 4dB/109dB/113dB
percussive classical, close mic : 4dB/118dB/122dB
see discussion of these numbers at
http://groups.google.com/group/rec.audio.h...ource&hl=en
QUOTE(w1L50n @ Feb 21 2007, 07:57)
I've been trying to learn about all this stuff from reading and starting at the fundamentals. In this pursuit I read all 17 pages....
Until a couple of posts ago, there was something that was confusing me:
'don't confuse physical loudness with dynamic range...both in db'
Thanks for that!
eeesh...17 pages...that was a hard fought piece of important intel however...now I can continue learning.
What is "physical loudness"?
Loudness, technically, refers to the intensity of the signal AS PERCIEVED BY THE LISTENER. It is a physiological variable, not a physical variable.
Intensity is the "SPL" or what-have-you.
sorry, shouldn't have replied with my 17 nanoseconds of lunch break. 
Yes, SPL, not physical loudness.
Anyway - I was answering the original question, not trying to summarise the entire thread of arguments. The end listener is fine with 16 bits, but it doesn't hurt to have more bits. More bits seem to sell more units of whatever hardware you're trying to sell
Yes, SPL, not physical loudness. Anyway - I was answering the original question, not trying to summarise the entire thread of arguments. The end listener is fine with 16 bits, but it doesn't hurt to have more bits. More bits seem to sell more units of whatever hardware you're trying to sell
QUOTE(krabapple @ Feb 21 2007, 19:14)
Fair discussion. "Audience noise, 13dB eqv" is so laughable that I can't believe anyone can take it seriously.
In a very nice large and well insulated anechoic chamber, where the one compromise was having to leave the (very large, padded motorised) door open by a few inches, I couldn't get lower than 20dB SPL A-weighted. Very few people can, anywhere. Never mind in a concert hall with an audience!!!.
Cheers,
David.
First post for me on here, and I guess reading the 17-page + argument is quite enlightening. There are obviously a lot a widely varied opinions on these topics, and although perhaps not backed by instrumentation and a lab environment, would like to report some of my own findings, which have taken a slow poke like myself years to really understand.
Let me state right upfront that I have been involved in recording, production, mixing and DJ activities for over a quarter of a century. Somehow this means experience, but also probably hearing damage, LOL!
The scope of my comment should be relating to one thing in particular: size of listening environment.
Here is the analogy: If I look at this site's graphic logo in its intended resolution for example, it looks perfectly fine, smooth and sharp. However, if I suddenly take the same logo and blow it up to fit a 2.5 meter-wide billboard poster (approx. 10 ft) then its edges will look ugly, jagged and just plain wrong....
My experiences have afforded me the opportunity to work on a weekly basis with very large-scale sound systems, the range going from 5,000 to 60,000 Watts of amplification, in rooms that can sometimes accommodate as much as 10,000 people.
The one thing which has become immediately apparent to me is that the bigger the room's volume and its sound system, the more easily I was able to distinguish artifacts which were totally inaudible in a smaller space. I also have access to studio-grade gear in vast quantity, and must confess that it is really hard for me to pick out the kinks between an MP3 file and its 16-bit 44.1 kHz source when listening on a pair of big Tannoy monitors and Bryston mono blocks in a studio, it seems that the sound fails to develop enough to really make a difference. Forget about headphones.
But surely I have done a number of tests on very large-scale listening systems, and that is where it has quickly become apparent that all of the extra information contained in a properly-encoded 24-bit / 96 kHz file does help make a noticeable difference in how it sounds compared to a 16-bit / 44.1 sibling.
Also experimented with up-sampling existing CD's via open-reel recording, simultaneously re-recording the result into a workstation at 24/96 to amazing results on quick A/B tests, which no doubt can be attributed to tape compression and other usual artifacts.
But there is no question in my mind that many of the arguments presented in the thread need to be illustrated within certain parameters, and that in my mind room size happens to be one of the most important ones.
The rough description of the type of tests I am basing my experiences on would be as below, in a room that holds about 1,500 people, roughly 30 meters wide by 25 meters deep with a ceiling height of 5 meters, with adequate amplification to bring sound to about 105 dB (?) without audible distortion, would imagine in the neigborhood of 15,000 Watts with plenty to spare, barely hitting '3' on the master volume:
-1) CD Red-Book Audio version of song DAC from Pioneer CDJ-1000 Mk 2 player.
-2) vinyl version of same. (obviously, depending on cartridge, tonearm calibration, etc.. used on test was standard DJ-grade Ortofon cart + Technics SL-1200 turntable)
-3) File up-sampled via 15 ips tape recording to 24/96 from workstation. AD from Metric Halo 2882 - DAC from RME Multiface.
-4) recording of vinyl at 24/96. AD from Metric Halo 2882 - DAC from RME Multiface
I can hear that the tape transfer has more bottom and smoother highs in pretty much any size room. Never mind that part, it's the 'tape-effect' for sure.
The big story for me was is how fond of COHERENCE my ears were.
When it comes to pure ear-pleasing power (punchiness, coherence and smooth presentation) the vinyl usually won as long as mastering wasn't horrible, even though there were many high frequency bits just missing in the very high upper register, and many percussive details in the midrange also a bit smeared.
The Red-Book audio CD sounded edgy and generally grating, screechy hi-hats, jagged brass and typical lack of bass, including what I'd call no 'woof' (percussive bass factor?), but certainly had a fair amount of extra detail and pleasant overall imaging/separation compared to vinyl.
The vinyl transfer to 24/96 PCM audio was certainly subjectively much nicer sounding than its CD counterpart, but obviously subject to the same limitations as its vinyl source.
The up-sampled tape to 24/96 seemed to have the best of all worlds: the cleanliness of its original CD source, along with the 'glue' that passing things through tape seem to bring, overall effortlessly smooth sound, and the kind of dynamic range that really made listening to the music a joy.
I realize that I should have included 16-bit / 44.1 kHz versions of the tape up-sample and vinyl transfers to really make this more of a valid test, but this is all I did. Not having access to DSD recording gear, this was obviously not included in the test. (let me not fail to thank Sony for its continued lack of support for the very formats it invents)
Quite empirical, for sure, but at least it left me convinced that there is a place for all of those extra bits, and that somehow judicious analog re-mastering can improve just about any sterile-sounding 16-bit digital source.
The last bit I really noticed is that when passing the pure digital source through a tube line amp, (didn't have the luxury to try a stand-alone tube D/A where I did this) it predictably smoothed things a bit more, and made the Red-Book CD at least listenable. The up-sampled 24/96 via tape sounded absolutely glorious through tubes on playback.
My little moral of the story also seems to be that anything that 'reconstitutes' the signal on playback, even if totally adding distortion and other artifacts appears to be the real important factor to pleasing the human ear in large-scale listening situations. The bigger the space, the more you notice it. For most people's everyday home listening situations, all of this stuff would probably make no difference whatsoever.
There you go, I hope some of you can see beyond my probable failure to follow some of the site's TOS for the bit of interesting data it may hold for those who care about its applications in similar large acoustic spaces.
I would of course welcome hearing any comments on this...
D.
Let me state right upfront that I have been involved in recording, production, mixing and DJ activities for over a quarter of a century. Somehow this means experience, but also probably hearing damage, LOL!
The scope of my comment should be relating to one thing in particular: size of listening environment.
Here is the analogy: If I look at this site's graphic logo in its intended resolution for example, it looks perfectly fine, smooth and sharp. However, if I suddenly take the same logo and blow it up to fit a 2.5 meter-wide billboard poster (approx. 10 ft) then its edges will look ugly, jagged and just plain wrong....
My experiences have afforded me the opportunity to work on a weekly basis with very large-scale sound systems, the range going from 5,000 to 60,000 Watts of amplification, in rooms that can sometimes accommodate as much as 10,000 people.
The one thing which has become immediately apparent to me is that the bigger the room's volume and its sound system, the more easily I was able to distinguish artifacts which were totally inaudible in a smaller space. I also have access to studio-grade gear in vast quantity, and must confess that it is really hard for me to pick out the kinks between an MP3 file and its 16-bit 44.1 kHz source when listening on a pair of big Tannoy monitors and Bryston mono blocks in a studio, it seems that the sound fails to develop enough to really make a difference. Forget about headphones.
But surely I have done a number of tests on very large-scale listening systems, and that is where it has quickly become apparent that all of the extra information contained in a properly-encoded 24-bit / 96 kHz file does help make a noticeable difference in how it sounds compared to a 16-bit / 44.1 sibling.
Also experimented with up-sampling existing CD's via open-reel recording, simultaneously re-recording the result into a workstation at 24/96 to amazing results on quick A/B tests, which no doubt can be attributed to tape compression and other usual artifacts.
But there is no question in my mind that many of the arguments presented in the thread need to be illustrated within certain parameters, and that in my mind room size happens to be one of the most important ones.
The rough description of the type of tests I am basing my experiences on would be as below, in a room that holds about 1,500 people, roughly 30 meters wide by 25 meters deep with a ceiling height of 5 meters, with adequate amplification to bring sound to about 105 dB (?) without audible distortion, would imagine in the neigborhood of 15,000 Watts with plenty to spare, barely hitting '3' on the master volume:
-1) CD Red-Book Audio version of song DAC from Pioneer CDJ-1000 Mk 2 player.
-2) vinyl version of same. (obviously, depending on cartridge, tonearm calibration, etc.. used on test was standard DJ-grade Ortofon cart + Technics SL-1200 turntable)
-3) File up-sampled via 15 ips tape recording to 24/96 from workstation. AD from Metric Halo 2882 - DAC from RME Multiface.
-4) recording of vinyl at 24/96. AD from Metric Halo 2882 - DAC from RME Multiface
I can hear that the tape transfer has more bottom and smoother highs in pretty much any size room. Never mind that part, it's the 'tape-effect' for sure.
The big story for me was is how fond of COHERENCE my ears were.
When it comes to pure ear-pleasing power (punchiness, coherence and smooth presentation) the vinyl usually won as long as mastering wasn't horrible, even though there were many high frequency bits just missing in the very high upper register, and many percussive details in the midrange also a bit smeared.
The Red-Book audio CD sounded edgy and generally grating, screechy hi-hats, jagged brass and typical lack of bass, including what I'd call no 'woof' (percussive bass factor?), but certainly had a fair amount of extra detail and pleasant overall imaging/separation compared to vinyl.
The vinyl transfer to 24/96 PCM audio was certainly subjectively much nicer sounding than its CD counterpart, but obviously subject to the same limitations as its vinyl source.
The up-sampled tape to 24/96 seemed to have the best of all worlds: the cleanliness of its original CD source, along with the 'glue' that passing things through tape seem to bring, overall effortlessly smooth sound, and the kind of dynamic range that really made listening to the music a joy.
I realize that I should have included 16-bit / 44.1 kHz versions of the tape up-sample and vinyl transfers to really make this more of a valid test, but this is all I did. Not having access to DSD recording gear, this was obviously not included in the test. (let me not fail to thank Sony for its continued lack of support for the very formats it invents)
Quite empirical, for sure, but at least it left me convinced that there is a place for all of those extra bits, and that somehow judicious analog re-mastering can improve just about any sterile-sounding 16-bit digital source.
The last bit I really noticed is that when passing the pure digital source through a tube line amp, (didn't have the luxury to try a stand-alone tube D/A where I did this) it predictably smoothed things a bit more, and made the Red-Book CD at least listenable. The up-sampled 24/96 via tape sounded absolutely glorious through tubes on playback.
My little moral of the story also seems to be that anything that 'reconstitutes' the signal on playback, even if totally adding distortion and other artifacts appears to be the real important factor to pleasing the human ear in large-scale listening situations. The bigger the space, the more you notice it. For most people's everyday home listening situations, all of this stuff would probably make no difference whatsoever.
There you go, I hope some of you can see beyond my probable failure to follow some of the site's TOS for the bit of interesting data it may hold for those who care about its applications in similar large acoustic spaces.
I would of course welcome hearing any comments on this...
D.
I will freely admit to having roughly a millionth of the production experience that you have, but from what I would expect, wouldn't the listening environment for a PA system involve far more distortion than for a high-quality nearfield monitor, or high-quality headphones? At least from a THD/frequency response point of view?
If so, that points to a highly theoretical and abstract mechanism of audibility. Extremely large variations in frequency response (and THD) could do crazy things with the ATH, so that certain changes to the sound that are inaudible with a perfect audio system become audible in distorting system. The "usual" example around here is that certain LAME encoder settings are quite transparent when using stereo material, but are substantially worse on Dolby Pro Logic stuff. Another example is that the presence of even order harmonic distortion could increase the audibility of absolute polarity reversal.
That said, unless your room with a 15kw PA system has a +40db high-Q peak somewhere, or intermodulates 10-100% at ultrasonic frequencies, I'm a little skeptical of the testing approach. Intuitively, the numeric distortions needed to bring high-res PCM accuracy to the audible range would have to be a little wild.
Out of curiousity, do you know the mastering changes between the the CD and vinyl releases of this record?
If so, that points to a highly theoretical and abstract mechanism of audibility. Extremely large variations in frequency response (and THD) could do crazy things with the ATH, so that certain changes to the sound that are inaudible with a perfect audio system become audible in distorting system. The "usual" example around here is that certain LAME encoder settings are quite transparent when using stereo material, but are substantially worse on Dolby Pro Logic stuff. Another example is that the presence of even order harmonic distortion could increase the audibility of absolute polarity reversal.
That said, unless your room with a 15kw PA system has a +40db high-Q peak somewhere, or intermodulates 10-100% at ultrasonic frequencies, I'm a little skeptical of the testing approach. Intuitively, the numeric distortions needed to bring high-res PCM accuracy to the audible range would have to be a little wild.
QUOTE
-1) CD Red-Book Audio version of song DAC from Pioneer CDJ-1000 Mk 2 player.
-2) vinyl version of same. (obviously, depending on cartridge, tonearm calibration, etc.. used on test was standard DJ-grade Ortofon cart + Technics SL-1200 turntable)
-3) File up-sampled via 15 ips tape recording to 24/96 from workstation. AD from Metric Halo 2882 - DAC from RME Multiface.
-4) recording of vinyl at 24/96. AD from Metric Halo 2882 - DAC from RME Multiface
-2) vinyl version of same. (obviously, depending on cartridge, tonearm calibration, etc.. used on test was standard DJ-grade Ortofon cart + Technics SL-1200 turntable)
-3) File up-sampled via 15 ips tape recording to 24/96 from workstation. AD from Metric Halo 2882 - DAC from RME Multiface.
-4) recording of vinyl at 24/96. AD from Metric Halo 2882 - DAC from RME Multiface
Out of curiousity, do you know the mastering changes between the the CD and vinyl releases of this record?
"The last bit I really noticed is that when passing the pure digital source through a tube line amp, (didn't have the luxury to try a stand-alone tube D/A where I did this) it predictably smoothed things a bit more, and made the Red-Book CD at least listenable. The up-sampled 24/96 via tape sounded absolutely glorious through tubes on playback."
Which from my (admitted) laymens knowledge means, you prefer the sound of higher harmonic distortion.
Which from my (admitted) laymens knowledge means, you prefer the sound of higher harmonic distortion.
QUOTE(mdmuir @ Apr 8 2007, 10:00)
"The last bit I really noticed is that when passing the pure digital source through a tube line amp, (didn't have the luxury to try a stand-alone tube D/A where I did this) it predictably smoothed things a bit more, and made the Red-Book CD at least listenable. The up-sampled 24/96 via tape sounded absolutely glorious through tubes on playback."
Which from my (admitted) laymens knowledge means, you prefer the sound of higher harmonic distortion.
Specifically, even order harmonics. Only tube amps have this. Not a 100% sure, so don't take this as a fact.
Here is some food for thought: http://www.meridian-audio.com/w_paper/Coding2.PDF
...though since it doesn't agree with what seems to be the stipulated view here, I guess it will be attacked on all fronts
And no, I don't think 16 bits are enough.
And yes, I think 192kHz is total overkill.
...though since it doesn't agree with what seems to be the stipulated view here, I guess it will be attacked on all fronts
And no, I don't think 16 bits are enough.
And yes, I think 192kHz is total overkill.
When you have the same signal being reproduced through multiple speakers at different distances from the listener, obviously you will have very complex combining of the audio, increasing some frequencies and canceling others. Clearly the normal psychacoustics of a lossy encoder are not meant to compensate for this.
As for the OP's other observations, I have little interest in how adding this or that distortion makes the sound more or less pleasing to him. And comparing vinyl to CD of the same material when nothing is known about how the two were mastered is pointless.
My interest in this forum was and continues to be how lossy codecs can be made to reproduce stereo material played in a normal listening environment with the least audible difference from the original.
As for the OP's other observations, I have little interest in how adding this or that distortion makes the sound more or less pleasing to him. And comparing vinyl to CD of the same material when nothing is known about how the two were mastered is pointless.
My interest in this forum was and continues to be how lossy codecs can be made to reproduce stereo material played in a normal listening environment with the least audible difference from the original.
I agree totally - especially on the last sentence.
QUOTE(tommypeters @ Apr 8 2007, 18:35)
Here is some food for thought: http://www.meridian-audio.com/w_paper/Coding2.PDF
...though since it doesn't agree with what seems to be the stipulated view here, I guess it will be attacked on all fronts
And no, I don't think 16 bits are enough.
And yes, I think 192kHz is total overkill.
That paper's been linked to several times on HA in the past few years.
Now, find us the published double blind listening tests results that support Stuart's assertions.
QUOTE(Madman2003 @ Apr 8 2007, 03:10)
QUOTE(mdmuir @ Apr 8 2007, 10:00)
"The last bit I really noticed is that when passing the pure digital source through a tube line amp, (didn't have the luxury to try a stand-alone tube D/A where I did this) it predictably smoothed things a bit more, and made the Red-Book CD at least listenable. The up-sampled 24/96 via tape sounded absolutely glorious through tubes on playback."
Which from my (admitted) laymens knowledge means, you prefer the sound of higher harmonic distortion.
Specifically, even order harmonics. Only tube amps have this. Not a 100% sure, so don't take this as a fact.
Depends on the kind of tube amp. Single ended amps have primarily 2nd order, with all orders present. Push-pull amps have all even-order harmonics very nearly cancelled, it's almost all odd-order distortions.
LP's have very strong assymetric (i.e. even order and odd order both) nonlinearities.
[/quote]
Most studios know that recording at > 44.1 kHz is pretty useless, unless it is a recording for DVD-A or SACD purposes. All studios I know record simply at 44.1 kHz / 24 bit. There are no artists that I know of that record at 192 kHz to give the end user a fuzzy feeling
[/quote]
OK so with my home recording gear limitations I basically have two options:
44.1/16 bit or 48/24 bit
Which would you opt for and why?
QUOTE(nrand @ May 1 2008, 07:20)
QUOTE
Most studios know that recording at > 44.1 kHz is pretty useless, unless it is a recording for DVD-A or SACD purposes. All studios I know record simply at 44.1 kHz / 24 bit. There are no artists that I know of that record at 192 kHz to give the end user a fuzzy feeling
OK so with my home recording gear limitations I basically have two options:
44.1/16 bit or 48/24 bit
Which would you opt for and why?
If it is going straight to CD as-is then 44.1/16. If you plan to do any kind of manipulation first then 48/24.
Edit: fixed quites.
The video world works at 48kHz.
If you were just planning on dealing with CDDA, it would be better to work at 44.1/24 than 48/24 (EDIT: if given the option), no?
If you were just planning on dealing with CDDA, it would be better to work at 44.1/24 than 48/24 (EDIT: if given the option), no?
Linn Records offer 24bit 96Khz recordings
http://www.linnrecords.com/linn-help-downl...-downloads.aspx
A lot of concerts on DVD have a 24bit 48 or 96 khz LPCM stereo audio track.
Is this just marketing or maybe when you need to do some post manipulation or processing it degrades the sound less at higher bit rates prior to downsampling.
I have done some AB comparisions with 24/96 vs 16/44.1 and the difference are subtle on certain music and not noticable at all on other music.
http://www.linnrecords.com/linn-help-downl...-downloads.aspx
A lot of concerts on DVD have a 24bit 48 or 96 khz LPCM stereo audio track.
Is this just marketing or maybe when you need to do some post manipulation or processing it degrades the sound less at higher bit rates prior to downsampling.
I have done some AB comparisions with 24/96 vs 16/44.1 and the difference are subtle on certain music and not noticable at all on other music.
Was this AB testing or ABX testing? ABX testing eliminates the placebo effect. AB testing doesn't.
Cheers, Slipstreem.
Cheers, Slipstreem.
QUOTE(Slipstreem @ May 2 2008, 16:05)
Was this AB testing or ABX testing? ABX testing eliminates the placebo effect. AB testing doesn't.
Cheers, Slipstreem.
This was ABX, with 2 people accessing the differences and a 3rd changing source.
Was using a wonderfull recording on David Gilmore Live DVD which has a 24/96 LPCM stereo of the concert. Ripped from the DVD and converted to 16/44.1 using Cakewalk and left as 24/96. It's one of the best recordings i have ever heard, highly recomended. Started me off in the interest of looking for LPCM music off DVD's.
Fair enough. I guess that the point of my post was to remind you to state clearly when ABX testing has been carried out in preference to AB testing. This forum can be a foreboding place to a newcomer so I thought I'd nudge you politely before any less forgiving members 'stuck the boot in', so to speak. We like to deal in certifiable facts. 
Cheers, Slipstreem.
Cheers, Slipstreem.
QUOTE(andrew & david @ May 2 2008, 11:20)
QUOTE(Slipstreem @ May 2 2008, 16:05)
Was this AB testing or ABX testing? ABX testing eliminates the placebo effect. AB testing doesn't.
Cheers, Slipstreem.
This was ABX, with 2 people accessing the differences and a 3rd changing source.
You also need to provide numerical results to show that it is statistically significant.
QUOTE(pdq @ May 2 2008, 17:56)
QUOTE(andrew & david @ May 2 2008, 11:20)
QUOTE(Slipstreem @ May 2 2008, 16:05)
Was this AB testing or ABX testing? ABX testing eliminates the placebo effect. AB testing doesn't.
Cheers, Slipstreem.
This was ABX, with 2 people accessing the differences and a 3rd changing source.
You also need to provide numerical results to show that it is statistically significant.
Also, were the sources level-matched within 0.1 dB? And properly time-aligned?
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