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?
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.
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.
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.
"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.
"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.
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.
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, 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.
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.
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.
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.
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.
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.
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