is it correct to assume that 24/96 audio would be indistinguishable from cd quality when listened to with speakers with a 20khz 3db and rapid hi frequency roll-off?
Or more precisely, that the only benefit comes from the shift from 16 to 24 bit, not the increased sample rate, as they higher freq content is filtered out anyhow?
related to this, which advice would you have for sub $5k speakerset with good higher freq capabilities for 24/96 audio?
Supposedly, the biggest advantage to 24/96 or 24/192 for that matter is the 24-bit data word, which allows a much lower noise floor, far more headroom during recording, and finer-grained loudness level tracking. To be honest I've never a difference between CDs and SACDs on my system, and I'm not sure the difference is audible. I have no doubt that for recording the extra headroom is a big advantage, but for playback I'm not convinced.
The higher sampling rates enable the use of less intrusive digital filtering circuits (better time-domain accuracy), which could make playback more accurate in the 20-20KHz bandpass, and maybe I just haven't heard the right equipment yet. Dinosaur that I am, I still haven't heard anything, from any source type or signal, that sounds better than my Levinson No39 CD player.
Got any 24/96 files around to listen to? Im guessing no.
Then go get some.
The High res Highway of files delivered natively at 16/48 (DVD) 24/88 (download), 24/96 via DVDA or Blu ray, or 24/192 via DVDA or BR again, or by downloads, are merely factions of the orig masters and how they are being presented.
The word lengths and sampling rates of the music (signal) do not translate to the frequencies we hear. They are merely methods. Formats. A means to an end.
Nominally, higher word lengths and sampling frequencies relate to better resolution, details, and ambient info and the ability of a system to retrieve it.
Download some for gfree they are out there. Have a sound card or DAC and a media player capable of playing them in bit perfect or bit true mode, and see for yourself.
Then you question should be what speaker in the $5K range best suits my ?
and not what speaker best suits better musicl information.
Actually I have listened to multiple HD tracks. The reason why I asked the questions is because I have actually downloaded from linn records or HD tracks multiple files, but i'm just not capable to hear the difference between 24/96 and 24/192, and to be honest, I think I hear some difference between 16/44 and 24/96 (from the same audio track on hdtracks) but less than I would have hoped. This is why I wondered if my speakers might need upgrading (as my spendor 3db point is at 20khz). My system consists of PC with USB or NAS, W4S DAC, LFD NCSE, Spendor s8e.
I agree with your final comment. My question was more a theoretical one as i try to understand what is going on from a processing/ technology point of view.
Ive said for a long time now and it should continue to be stated . If you do not hear a diff . then dont pay for it.
Im not familiar with your speakers.
I am familiar with some things about human hearing though. Your speakers will exceed your ability to hear what they can reproduce.
Remember, dont confuse the numbers. Bit rates and sampling frequencies are parts of the mastering/recording/editing production process.
We hear in an analog world not in the digital realm.
We dont hear those hyper cycles. Many of us here only hear below 16KHz. Or less. Many dont hear well below 70 Hz or so distinctively. Without a test disc nearby I couldnt tell you if a note was 50Hz or 60Hz. I could only say one is deeper/lower than the other. Das eet! If Im paying attention at the time!
No one thing usually in a system dictates the systems overall voice its a en masse affair. That said, Id look into each area of my outfit to increase resolution if that is what I wanted to do.
Primarily Id look for gains in my source (s). Resolution and details combine to reconstruct the recorded venue in our spaces or bring us to those artists, better. Without a great source unit producing those pure signals with all that info within them, not too much thereafter will get that info back for you.
But it all makes a difference for sure components cabling, conditioning, amplification, and of course, speakers.
Thereafter, as you appear to be using the pc as a source, which media player is in use? What output in that player is selected? Which driver/engine is decoding the info? If you are using USB are you using an ASIO driver? Is your DAC capable of handling 24/96 over USB? Are you certain you are getting bit perfect output?
Answer those Qs positively and Id say either stay where you are at, or begin by upgrading the DAC or the interface the DAC uses out of the PC. Switch to BNC for example. BNC has no limitations as does some USB DACs.. Albeit most DACs today handle 24/96 via USB fairly readily . Albeit some do not. Mine does not. It only does Red Book over USB.
Once you are sure you are getting bit true input to your DAC at the proper sampling rates and bit depths and the DAC is processing them right you should be able to hear a diff How much of one again depends on YOU, your room, and your outfit.
BTW some DACs do very well indeed at this rate or that, but show some lack at other rates or via other interfaces. Mine for ex likes AES, then BNC, then coax/RCA, then its a toss up between USB and Optical.
Even your aSio DRIVER can be a game changer . Depending on which one you use . So too can be the USB cable.
But as you seem now to be in the market for spakers . Remember, it aint just the speakers most likely . Id look upstream and review the above Qs.
Mizuno - in order to avoid aliasing there should be no signal at 1/2 of the sampling frequency. In order to achieve it data has to be filtered out at 1/2 of sampling frequency in A/D processing.
Notice, that we are talking about preserving frequency information only (no aliases). Amplitude wise 16/44.1 will be very limited. Lets assume that you can hear 15kHz. Make picture of one full cycle of sinewave on a paper and try to place 3 points on it (reconstruct with 3 points only). You see the problem. Second problem is that filtering out info above 22.05kHz requires steep filters. Steep filters time shift different frequencies by different amount (uneven group delays) making inaccurate summing of harmonics. This will also screw-up step response (transients). Steep filters are not used in SACD recording making step response better. Of course master tapes are recorded in higher rate and re-sampled down but 96kHz playback will be still better than 44.1kHz (more points). 192kHz contains even more points but playback at 192kHz is not necessarily better than at 96kHz where THD of the most D/A ICs is the lowest (unless DAC uses extra info - downsampling). Resolution wise 24bit is better but most of converters are limited to about 20 bits anyway. Traditional converters are limited by tolerance of components to about 18 bits while Delta-Sigma are limited by timing errors to about 20 bits. One possible exception is Ring-DAC used by DCS (and previously licensed to ARCAM) that gets extra resolution by switching identical components of divider ladder in order to obtain more accurate average value. Some of the resolution will get buried in system noise that comes either from jitter (noise in time domain)or power amp's S/N.
Kijanki, your example of a 15KHz sine wave and "three points", implying poor reproduction, isn't the case, according to the well-proven Nyquist Theorem. 20KHz is reproduced as accurately as 1KHz with a 44KHz sampling rate. As for the phase shifts from steep digital filters, this is what over-sampling was invented to address, and eases the difficulty of designing a good anti-aliasing filter. For playback 16/44 is probably better than your audio system. (As I mentioned, for recording you might want the headroom of a longer word-length.)
Bob R is correct about some amps being a limiting factor in hearing the better s/n ratios of longer word-lengths. Most high-end amps are rated as having noise levels about 100db below full power. Since amps usually have about 25-27db of gain that means that their s/n ratio is only about -75db at 1 watt, or well inside the capabilities of 16/44. Krell amps, for example, are about the best, and they have s/n ratios in the mid-high -80db range at 1 watt.
Irvrobinson - 20kHz frequency is reproduced accurately (no aliasing). As for the amplitude, theorem assumes infinite number of samples (of periodic signal). Because it is not the case, interpolation is done with Sinc functions but with constantly changing signal that is close to 1/2 of sampling frequency it is very coarse. More samples would be better IMHO.
As for oversampling in A/D process - even if you sample at 192kHz your filters have to get 96dB attenuation at 96kHz to be 16-bit perfect. Such Bessel filter would have to have perhaps 16 or so poles. Attenuation of 20kHz/-3dB 8 pole Bessel filter is only 50dB at 96kHz. Fortunately signals at 20kHz have very low amplitude so that might be OK.
I like 16/44 and agree that a lot can be improved in other areas. Jitter, being source of noise, is one of them. We learned to remove jitter by better (dual) Phase Lock Loops or asynchronous rate converters (upsampling) but there is still some jitter from less than perfect A/D processing that cannot be removed (common for older recordings).
Kijanki - 20KHz reproduction with a 44KHz sampling rate is perfect for sine waves, not "coarse". A higher sampling rate doesn't improve accuracy within the frequency response of the lower rate, it just extends the frequency response. That doesn't mean I think digital recording and reproduction is perfect overall, it just means that in terms of capturing the frequency domain information at 20KHz, 44.1KHz sampling is completely sufficient to perfectly capture the sine waves. I think people confuse digital sampling with analog interpolation, and it isn't similar.
"capturing the frequency domain information at 20KHz, 44.1KHz sampling is completely sufficient to perfectly capture the sine waves"
Maybe sufficient for sinewaves but not for the music because it would call for brick wall filters that have very uneven group delays (non-linear phase if you prefer) and will cause wrong summing of harmonics. Such setup will be OK for single frequency reproduction but will be very unpleasant with music (dynamic signal).
Yes it is coarse because Nyquist-Shannon theorem requires infinite amount of terms (samples). Fixing it with sin(x)/x works poorly for short bursts around 1/2 of the sampling frequency. Sound of instruments producing continuous sound might be not affected (like flute) but anything with transients will sound wrong (piano, percussion instr. etc). Notice, that when people compare analog to 16/44 first thing they notice is different sound of the cymbals.
On the other hand, if you still think it is perfect system - enjoy.
"Maybe sufficient for sinewaves but not for the music because it would call for brick wall filters that have very uneven group delays (non-linear phase if you prefer) and will cause wrong summing of harmonics. Such setup will be OK for single frequency reproduction but will be very unpleasant with music (dynamic signal)."
I have no idea what you're talking about. The wrong summing of harmonics, and it'll be very unpleasant? I don't know about that, Kijanki. You talk like a technically competent person, but then you make these outlandish claims. So if these summed harmonics will be so screwed up, why is it that those of us with very good high frequency hearing and high quality speakers can't hear anything very unpleasant? And if they do sound so unpleasant, why when I listen to higher res stuff through a Benchmark DAC it doesn't sound noticeably better?
I think you're exaggerating the issues, and wrapping the arguments in technical-sounding reasons that really don't alter the music audibly.
I was trying to show that 16/44 recording wouldn't be a perfect process and that's why it is done in 24/192 but downsampling to 16/44 also takes away quality.
Digital reproduction (as well as analog) have limitations. Filtering screws up transient response and 16 bit resolution is less than perfect.
Why it is difficult to hear difference thru Benchmark? Possibly because available hi-res is often poorly made (many complains about that) while our systems and rooms have shortcomings.
Power amp might be limiting factor but it isn't as bad as Irvrobinson calculated. First of all S/N or THD+N of the amp is usually shown at 1W and many amps are better than 96dB. In addition we don't listen at 1W . For instance if we take Rowland 625 amp's S/N specification of 95dB at 1W at 8 ohm it will be higher at the output power of 300W. We might look as well at residual output noise specified by Rowland that is 55uV at 20Hz-20kHz unweighted. Since output voltage at nominal power of 300W is around 50V it makes S/N=119dB. SACD reproduction is roughly equivalent to 20/96 requiring dynamic range of 120dB. D/A converters are also limited to 20 bits performance.
So to answer original question - increasing resolution might be beneficial up to about 20 bits assuming good recording/file, system and room. Increasing rate will be always beneficial to avoid serious shortcomings I mentioned before.
I settled at standard redbook reproduction not only for practical reasons but also because I cannot stand hiss and pops of analog playback that don't allow me to forget I'm not sitting "there" at the concert.
And if they do sound so unpleasant, why when I listen to higher res stuff through a Benchmark DAC it doesn't sound noticeably better?
You can buy Tom Petty Mojo in CD or in HD and compare. There is a difference but most of the difference is due to audio compression applied to the CD master to make it "hot" - see CD loudness wars and what artists and producers try to do make the music to make it sell.
Basically they compress everything - especially drums - so that the dynamic range of peaks above RMS is usually no more than 6 to 10 db. Whereas a good recording in pop/rock may have 20 db peaks and a classical recording may have 30 db peaks above RMS.
The HD files - such as those on HD tracks are usually much less compressed than the 16/44.1 equivalents.
To add to my last comment...from a purely "technical" perspective I would agree that CD quality is more than adequate as a playback medium. The problem is NOT the CD media itself but more a problem with what the producer and mastering do to the music BEFORE it gets issued as a CD.
Of course 24/48 or 24/96 is essential in the Studio because there is much more dynamic range and signal manipulation required in that environment.
Shadorne, agree about compression but it is simply who is driving the market. Release uncompressed piano recording (about 96dB dynamics) and a lot of people will complain that on their boom boxes or shelf speakers woofers are constantly buzzing. Hi-res has different clientele so they reduced compression a bit but it is still bad. Also, as you mentioned, they try to make average loudness as high as possible because to inexperienced customer it appears as higher quality recording especially with poorly resolving systems.
I think redbook CD format specifies the dynamic range for the 16 bit format and is fairl y standardized as a result.
Not sure this is the case with other newer hi rez digital formats?
More bits enables more dynamic range and more detail together. How this happens might be highly variable in lieu of a standard.
In any case, for hi rez digital sources, I suspect a difference associated mainly with the high frequencies can be heard if done right, but that may be a big if at this juncture still.
To hear the most possible, you definitely want very good, younger ears, speakers that can handle dynamics and transients well and also have very good detail assuming the production is done well and the DAC not only reads the format but is able to output analog of similar resolution and quality.
At this still emerging stage of hi rez digital audio, I doubt it is a safe bet that hi rez source material and playback gear meets these requirements well in general, although I am sure there is some reference type recordings and better gear that do.
The first place I would listen for the difference is in well recorded massed bowed strings in orchestral music. Use a good modern RTR reference recording as a reference standard. Even older trained ears should hear a noticeable difference if the digital is not extremely well done.
I have had the opportunity to listen to RTR, vinyl, and good redbook CD recordings on a very well done dealer system using mbl 111e speakers. The difference from RTR to redbook CD was pronounced but you might not notice the limitations of the redbook CD unless compared to the RTR or even good vinyl on a similarly good system.
The SOTA wide and deep soundstage in this optimized and very resolving mbl setup provided exactly the venue size and 3 dimensonal sound quality needed to be able to hear these kinds of differences clearly. Quite an eye (ear?) opener!
First of all, I totally agree about compression running rampant these days, especially for drum kits. My wife is a drummer, so I know what drums really sound like, and only a few recordings give you a hint of their dynamic range. In fact, she and I both lament that a lot of modern recordings don't even use real drums any more, only those electronic travesties, for ease of recording.
To expand about about differences I've heard with hi-res on the DAC1, sometimes I think I hear a difference, in that some hi-res recordings seem to reveal something I've never heard before, but then I go back to a CD and hear similar things. Or I find an awesome recording on CD that sounds better than anything I've heard on hi-res. For example, oddly, I still haven't heard a piano recording superior to the ancient Telarc CD of Malcom Frager playing Chopin. That old Soundstream recording even forces an odd conversion to 16/44, and it still sounds great.
As for Kijanki's comment that the s/n ratio of most amps is specified at 1W, I say check again. All of the obvious ones I've checked reference full power, and most can't break -85db at 1W into 8ohms. JA's measurements in Stereophile are very interesting in this regard. (His measurements are the only reason I read the magazine.)
Irvrobinson 6-30-11: I still haven't heard a piano recording superior to the ancient Telarc CD of Malcom Frager playing Chopin.
If you can find it, try Wilson Audio WCD-9129, Chopin's Piano Sonata No. 3 in B Minor, Op. 58 (and other shorter works), performed by Hyperion Knight. The best reproduction of solo piano in my experience, and it's on a 1991 redbook cd!
What can I say - I posted example showing that quality amplifier is not a limiting factor. Why not to respond to that? If you think you can find any mistake in my reasoning please say so. Even for my own Rowland 102 (a class D amp) dynamic range is stated as 110dB while Rowland 301 is rated 120dB. Every Krell is at least 106dB unweighted (Evolution 900e is 113dB unweighted related to full power). You can search for a bad amp but the point was to show that the amp is not the limiting factor.
As for the Benchmark DAC1 again - If you cannot hear the difference then you can not, but please don't bring Nyquist into discussion since his theorem was intended toward stationary waveforms (infinite number of samples). Closer you get to Nyquist frequency the more samples you need to properly reconstruct original waveform - not possible to do for short high frequency sounds. That is the main reason so many people still stay with vinyl (unless you think they like convenience).
BTW, I should have added to my previous post that I am in agreement with all of the technical points Kijanki has made, which I think have been very well presented. An additional point which I don't think has been mentioned is that brickwall anti-aliasing filters introduce some degree of ripple into the passband frequency response characteristics, as I understand it.
The audible significance of all of the effects that have been mentioned, though, is perhaps unanswerable in a definitive manner, given the extent to which those effects tend to be overshadowed by variability in recording engineering and quality.
Al, Thank you. You brought very important point - quality of the recording engineering (on the top of compression issue that Shadorne mentioned). Average quality is not very high while some of the recordings I have are just incredibly good. Perhaps I'm arguing too much for the best case scenario while average quality of the recording was another reason for me to stay with 16/44 and Benchmark DAC1.
I dunno, Kijanki, I randomly looked at two good power amps in JA's testing, a Moon and a Pass, and they were both had measured s/n ratios of about -84db at 1 watt, which is actually excellent performance. You keep forgetting that most power amps have about 30db of gain. Measuring s/n ratio at full power is sort of cheating for marketing's sake. A speaker with 95db/2.83v/1m efficiency, like my old Legacy Focus, will let you hear the hiss from such an amp rather readily. So I still contend that for listening the amps are the limiting factor, not well implemented 16/44.
As for your comments about Nyquist, it would seem your real thesis is that digital reproduction isn't very good, even with a DAC1. I still wonder, why does it sound so good if you're correct? I'm missing something.
Irvrobinson - I assume that you buy properly sized amp for the speakers and the room. My amp is rated 150W at 6ohm and I am pretty sure I am getting peaks even larger than that (headroom). It corresponds to largest digital number coming from CD - meaning covers full dynamic range. If you listen at 1W then I agree that you have no chance to experience full dynamic range, not only because of the noise floor of the amp but more likely because of the ambient noise and threshold of our hearing.
To test if power amp is limiting factor is very simple - Just turn on power amp, set volume to zero and listen. Can you hear anything? I cannot - dead silent. If I cannot hear anything in very quiet room in my listening position why even bring numbers into discussion?
As for Nyquist - digital reproduction is decent from 16/44 media and, according to reviews, pretty good with SACD. I seriously doubt that they would release 24/192 master tapes to public. What is released right know as high resolution is often the same as 16/44 (I read article about it). SACD is a different story because it cannot be copied (pit width modulation) but it does not work with the server and selection is very limited. I settled at 16/44 for all the reasons I mentioned before but understand its limitations. I adjusted my gear accordingly with very forgiving Hyperion speakers.
. Closer you get to Nyquist frequency the more samples you need to properly reconstruct original waveform - not possible to do for short high frequency sounds.
Not so. The waveform is perfectly reconstructed. The mathematics are quite rigorous. The main issue with digital is
1. Anti alias filtering (higher frequencies must be eliminated prior to ADC or they can fold in) 2. Jitter
Both of the above add spurious non musical signals. Both can be managed
On the S/N discussion, this is usually around 100 dB on good gear. I am certain this is achievable because my speakers can hit about 112 dB SPL at the listening position (12 feet back) as measured with a SPL meter whilst I cannot hear any sound (when no music is playing) from the tweeter unless my ear is within about 6 inches. This translates to roughly 100dB(taking into account the difference in distance which is around 12 dB and assuming the threshold for hearing hiss is around 20 dB in the room with inherent ambient noise around)
I think the ambient room noise and the speakers peak clean SPL are the limiting factors in a typical setup.
I think tape hiss or vinyl noise is limiting you to about 60 or 70 dB dynamic range on analog recordings.
I think high quality digital recordings can probably achieve around 90 dB dynamic range - limitations being the ambient noise during the recording process.
This is why CD is more than good enough for playback. This is why there are a few rebook CD recordings that are world class.
Of course, in a studio the signals are manipulated - this creates the need for even greater dynamic range (24 bit or 144 dB) - not that they will necessarily have better S/N but they may want to boost some sounds by 20 dB or so and may apply digital filters (the accuracy of said filters improves significantly if you have more bits)
07-01-11: Shadorne Of course, in a studio the signals are manipulated - this creates the need for even greater dynamic range (24 bit or 144 dB) - not that they will necessarily have better S/N but they may want to boost some sounds by 20 dB or so and may apply digital filters (the accuracy of said filters improves significantly if you have more bits)
Excellent point!
06-29-11: Kijanki ... Nyquist-Shannon theorem requires infinite amount of terms (samples). Fixing it with sin(x)/x works poorly for short bursts around 1/2 of the sampling frequency. Sound of instruments producing continuous sound might be not affected (like flute) but anything with transients will sound wrong (piano, percussion instr. etc).
06-30-11: Kijanki Closer you get to Nyquist frequency the more samples you need to properly reconstruct original waveform - not possible to do for short high frequency sounds.
07-01-11: Shadorne Not so. The waveform is perfectly reconstructed. The mathematics are quite rigorous. The main issue with digital is
1. Anti alias filtering (higher frequencies must be eliminated prior to ADC or they can fold in) 2. Jitter
Both of the above add spurious non musical signals. Both can be managed.
In theory Kijanki is correct. An infinitely long series of samples is required for the mathematics to work out perfectly. The consequences of that will be most significant for spectral components that are transient and that approach the Nyquist frequency (i.e., half the sample rate).
The extent to which that may be audibly significant on most recordings is probably conjectural. The Wilson Audio cd I referenced, among many others, leads me to believe that in general it is not a major factor as a practical matter.
Shadorne is of course correct, IMO, in emphasizing the significance of anti-alias filtering and jitter.
Bottom line: I am not losing any sleep over hi rez digital. There are too many ifs to really matter at this point for me and the benefits are marginal compared to the extra cost and overhead associated with even larger data files.
Al, I found video to show what happens when sampling just above Nyquist frequency. It might be possible to fix the output with sinc or other reconstruction functions but only if signal lasts a lot of cycles. If signal is short and disappears reconstruction will have huge error.
Thanks, Kijanki. The one thing I would question in your comment is the word "huge." I'm sure that a suitably chosen test waveform comprising a very short burst of high frequency energy, and put through a 44.1kHz a/d + d/a, can result in an error that will appear huge when viewed on an appropriate time scale. But as the saying goes the proof is in the pudding, and I've felt amazed at times at how good SOME cd's that contain a lot of transient high frequency energy can sound.
Al, Huge errors applied to the highest harmonics only will result only in small sound change. There will be small difference in sound of cymbals and perhaps in ambiance. I use 16/44 and like it, but try to be educated about it. That's all.
Bob - Thanks for the link. I suspect that THD is a dominating factor at higher power. Noise issue itself is non-existent in my opinion because if I cannot hear anything in a silent room at full power (dead silent) I don't worry. Many amps with similar 80dB THD+N performance are showing -120dB noise floor on the other graphs. Also, small amount of noise helps to increase resolution - technique known as dithering widely used in photography. I would be more concerned with THD and it doesn't look good.
I don't know what is relationship between THD and resolution but I suspect that resolution will still bring better sound. Another reason for that is quantization noise that is smaller at higher resolutions. DAC1 does very good job here by using sigma-delta converter that pushes quantization noise to higher bandwidth (oversampling).
I think that our hearing ability ends up slightly above 16-bit perhaps 18-20bits but I'm more concerned with sampling rate because low sampling rate in addition to phase shifts in steep low pass filters increases quantization noise (or size of square steps to make it simpler).
"I think that our hearing ability ends up slightly above 16-bit perhaps 18-20bits but I'm more concerned with sampling rate because low sampling rate in addition to phase shifts in steep low pass filters increases quantization noise (or size of square steps to make it simpler)."
It's statements like that, Kijanki, that make me wonder if you know what you're talking about. The width of the data word has nothing to do with our hearing ability. How many bits per word determines how many loudness levels there are. It's the sampling rate in KHz that determines how high the frequency response goes. You know that, right?
That video is wrong. It is showing a stair step signal which is NOT what the output of a DAC would look like. The output will be smoothed by a filter in order to eliminate all that horrible spurious high frequency signal from the stair steps. The output filter will remove the stair step and restore the sine wave so that the signals are much more alike - even a little above Nyquist - absolutely No need to got 100Hz sampling to properly render a 10Hz sine wave.
warning not everything you see from Universities is accurate.
Also it is WRONG to compare signals in this way. We hear frequencies NOT the waveform as presented graphically! The closeness of the waveforms as presented graphically is NOT a proxy for how close alike they will sound!
Shadorne - You have no clue. Filter will smooth-out the steps but will never remove few Hz modulation that was shown at 21Hz sampling rate. Al, please help me here or I'm going to kill myself.
Irvrobinson - I'm not talking about frequency range of our hearing but rather resolution of our hearing similar to number of shades of gray you can distinguish taking into consideration adverse conditions like ambient noise, system noise, THD, IMD etc.
I don't care anymore to defend myself from such attacks. You guys have no basic education in electronics and post nonsense just to keep arguing. Signing off.
Kijanki & Shadorne, you're both basically right but you're referring to different things.
Shadorne is alluding to the fact that a low pass reconstruction filter will smooth out the steps and restore an essentially perfect sine wave, if the original analog input was a sine wave at a frequency slightly less than the Nyquist rate (or lower). Of course, the filter itself may have significant side effects, but that is another subject.
Kijanki was alluding to the fact that if the analog input is a brief transient lasting for a limited number of samples and having spectral components approaching the Nyquist frequency, then the mathematics won't work out ideally no matter how ideal the reconstruction process is. Which is correct, although as I said earlier whether or not that may be audibly significant with worst case material (e.g., high frequency percussion) is probably a matter of conjecture. Admittedly, the video does not directly relate to Kijanki's point.
As far as the relation between low sample rates and quantization noise is concerned, while lower sample rates would obviously result in coarser steps in the sampled (unreconstructed) waveform, I think that Irv is basically correct to the extent that the reconstruction process can be accomplished ideally. However, given the possible effects on high frequency transients that we've been discussing, that may result from having a limited number of samples, and given the non-idealities of real-world filters, I suppose there could be some second-order relation between sample rate and quantization noise. It's been a long time since I took the relevant courses. :-)
Kijanki, are you implying that 24bit data words have a "finer grain" than 16bit data words? That each bit represents a smaller incremental signal level?
"Kijanki, are you implying that 24bit data words have a "finer grain" than 16bit data words? That each bit represents a smaller incremental signal level? "
That's the basic reason to us more bits in each sample in digital signal processing of any kind, isn't it?
Most sounds last at least a hundredth of a second or longer. My point is that even for a 15 KHz sound you are likely to be hearing 15000/100 = 150 cycles. It is irrelevant that the amplitude of a few cycles may not be graphically represented perfectly. The problem is the context we are talking about is related to hearing rather than graphical presentation of a waveform.
Although Kijanki is right about the graphical accuracy my point is that,as regards to human hearing and music, this is not so relevant. In essence the engineers at Sony and Philips did a thorough job when they came up with rebook CD! Perhaps if redbook CD was not as good as it is then SACD would not have failed. The problem is that SACD and other higher resolution formats are very much into diminishing returns compared to a well produced CD.
I would add that the graphical representation of waveforms and the "digital staircase" form one of the biggest and most enduring audiophile myths that analog is inherently better than digital. In fact, most of the benefits of analog come from the added distortion that is pleasing to the ear - analog tape machines are wonderful devices for audio compression(removing dynamic range)!
I think redbook CD done perfectly correctly both in recording and playback does fit the bill very well as designed as Shadorne indicated.
THe problem is more often the difference between design and theory and its realization in products, which is imperfect.
In order for hi res digital to make a difference, quality standards for accuracy have to be raised as well from end to end. To do that is relatively expensive still, I believe, though technology advances and should become more practical and affordable to achieve sometime down the road.
Not to say there may not be a practical advantage today for some, but this is very marginal at best, very expensive, and still probably not where I would want it to be in terms of technological maturity for me as a fairly average Joe audio buff to buy in.
I do need to download some hi res files sometime soon though and actually test out the waters a bit (no pun intended).
Shadorne, I am in essential agreement with your last post. As I said earlier:
An infinitely long series of samples is required for the mathematics to work out perfectly. The consequences of that will be most significant for spectral components that are transient and that approach the Nyquist frequency (i.e., half the sample rate).
The extent to which that may be audibly significant on most recordings is probably conjectural. The Wilson Audio cd I referenced, among many others, leads me to believe that in general it is not a major factor as a practical matter.
I particularly second your statement that:
... the graphical representation of waveforms and the "digital staircase" form one of the biggest and most enduring audiophile myths that analog is inherently better than digital.
BTW, the following excerpts from the technical notes accompanying the Wilson Audio cd I referenced (which I indicated provides the best reproduction of solo piano in my experience) may be of general interest:
The recorded perspective of the piano in this recording is close, as though the 9' Hamburg Steinway in being played for you in your living room. Of course the actual recording was not made in a living room! Instead, the great room of Lucasfilm's Skywalker Ranch, with its incredibly low noise floor and fully adjustable acoustics, was used.... A pair of Sennheiser MKH-20 omni microphones were employed ... amplified by two superb pure class-A microphone preamps custom-built for Wilson Audio by John Curl. MIT cable carried the balanced line level signal to Wilson Audio's Ultramaster 30 ips analog recorder. Subsequent digital master tapes were made through the Pygmy A/D converter on a Panasonic SV-3700.
Glad to hear we can all agree. Sony and Philips engineers did a great job with redbook CD, it would indeed be hard to go against all their research.
I agree that transients close to the Nyquist are going to be the most challenging to reproduce faithfully, however, there is really not much in th eway of sounds that one can call music above 15 KHz anyway.
"Kijanki, are you implying that 24 bit data words have a "finer grain" than 16 bit data words? That each bit represents a smaller incremental signal level? "
"That's the basic reason to us more bits in each sample in digital signal processing of any kind, isn't it?"
That isn't my understanding of how it works, Mapman. Each bit of word length corresponds to 6.02db of dynamic range, or, technically, s/n ratio. So, with 16 bit words you get about 96db of dynamic range, and with 24 bit words 144db. And, of course, 65,535 voltage levels with 16 bits, and 16,777,216 with 24bit words. So 24 bit is "finer grain"?
Yes and no. Yes, there are 16M levels, but there's so much more voltage range to cover. If you reference the maximum level to, say, the 2v max line level used in consumer audio, and that's 24 1s in a row, then all zeros will be 144db below 2v, which would take cryogenic circuits or whatever to achieve. That means you're wasting the bits below the resolution of modern amplification systems, which is probably something like 100db below 2v of power amplifier output, including all amplification stages (I'm being very generous), which means you're wasting 44db of dynamic range, or about 7 bits of word length. So that means you're probably using about 17 out of 24 bits in a real system. And, of course, I'm probably generous by several db of s/n ratio in a real system, so 16 bits isn't far off of what is the resolution limit in a home system, and "finer grain", meaning a better s/n ratio, won't be audible by most mortals.
What 24 bit words are good for is eliminating digital clipping in the recording studio. There's no such thing as a 144db peak in music. :-)
So unless I'm completely misguided (always possible) 24 bit audio isn't really "finer grain" in the 96db of dynamic range that 16 bit audio can encode. 24 bit just goes higher or lower, or a bit of both.
Irv, keep in mind that it is generally accepted that signal can be perceived at levels that are significantly below the level of random broadband noise that may accompany the signal. 15db or more below, iirc. So amplifier noise floor is not really a "floor" below which everything is insignificant.
Also, quantization noise is significantly correlated with the signal, at low signal levels, and is therefore perceived as distortion rather than noise. Dithering will minimize that effect, but it has its limitations and my understanding is that it is often not properly applied.
That said, I think we are all in agreement that the main usefulness of 24 bits is in the creation of the recording.
That said, I think we are all in agreement that the main usefulness of 24 bits is in the creation of the recording.
Agreed.
However, I would add that high resolution recordings are targeted at audiophiles - so this new high resolution media is useful in that you tend to get a better quality recording that has NOT been heavily compressed for mass consumption. So they ARE useful to audiophiles but not so much from the "improved resolution" but mostly because audio that gets formatted this way tends to be a better quality master rather than a master intended for restaurant, pub, iPod & car FM radio play.
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