speakers for 24/96 audio

Hi Blindjim,

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.

thx for the help :)

I’ve said for a long time now… and it should continue to be stated…. If you do not hear a diff…. then don’t pay for it.

I’m 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, don’t 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 don’t hear those hyper cycles. Many of us here only hear below 16KHz. Or less. Many don’t hear well below 70 Hz or so distinctively. Without a test disc nearby I couldn’t tell you if a note was 50Hz or 60Hz. I could only say one is deeper/lower than the other. Das eet! If I’m paying attention at the time!

No one thing usually in a system dictates the systems overall voice… it’s a ‘en masse’ affair. That said, I’d look into each area of my outfit to increase resolution… if that is what I wanted to do.

Primarily I’d 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 I’d 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 it’s 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 ain’t just the speakers most likely…. I’d look upstream and review the above Qs.

Good luck…

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).