research paper on high-def storage, AES 31rst Int. Conf. London Options

[boombaard]

i just was linked this article Which of the two Digital audio systems best Matches the Quality of the Analog System?

However, to achieve a higher degree of fidelity to the live analog reference, we need to convert audio using high sampling rate even when we do not use microphones and loudspeakers having bandwidth extended far beyond 20 kHz. Listeners judge high sampling conversion as sounding more like the analog reference when listening to standard audio bandwidth.

and this quote struck me as very, very odd.
(read the article first, or at least the conclusions/test setup bits before posting.. the phrasing seems to be a bit awkward in places, but i can't really help that)

because unless i'm really, really reading this wrong they're saying that if you upsample to the highest SR your sound card can take before playing it back - no matter the quality of the input material - it will be perceived as 'more natural'.

(mind you, i don't really see why you'd need to do what they suggest next, namely:

These results suggest that the archiving community should consider using high-sampling conversion to ensure transparency even if the recording is made with standard audio-bandwidth transducers, and when digitizing older recordings made with bandwidth-limited analog systems.

just turning on SSRC in your foobar dsp chain would seem to be enough)

the point, really, is that i almost can't believe that i'm understanding this right.
A second point: even if they are correct, it would seem that their conclusion that anything >20kHz is a waste of bw is somewhat hasty, and they imo should've added a 'middle' test - to say 40kHz - just to narrow down where the added bandwidth became overkill (more).

[Paulhoff]

Just because you write more does not mean you have more information. It just sounds like more audio woo-woo.

Paul

[boombaard]

Just because you write more does not mean you have more information. It just sounds like more audio woo-woo.

Paul

no offense, but your pointless (and content/meaningless) replies in this thread and the psy model one are starting to annoy me.
i suspect others as well, at that.. i'd suggest you stop posting unless you have something to contribute.

This post has been edited by boombaard: Dec 24 2007, 01:45

[Paulhoff]

Just because you write more does not mean you have more information. It just sounds like more audio woo-woo.

Paul

no offense, but your pointless (and content/meaningless) replies in this thread and the psy model one are starting to annoy me.
i suspect others as well, at that.. i'd suggest you stop posting unless you have something to contribute.

I guess you only read what you what to. Writing more data that has no more information is woo-woo.

Paul

[Light-Fire]

...because unless i'm really, really reading this wrong they're saying that if you upsample to the highest SR your sound card can take before playing it back - no matter the quality of the input material - it will be perceived as 'more natural'...

It is just the work of students of the MUSIC school of the University and NOT Engineering. That is why the "conclusions" sound (and are) absurd.
Look at the source before believing in something someone says. Besides. What site is this:http://www.hitech-projects.com?!

(mind you, i don't really see why you'd need to do what they suggest next, namely:

These results suggest that the archiving community should consider using high-sampling conversion to ensure transparency even if the recording is made with standard audio-bandwidth transducers, and when digitizing older recordings made with bandwidth-limited analog systems.

This is absurd. When trying to preserve some information we should avoid deterioration through upsampling.

This post has been edited by Light-Fire: Dec 24 2007, 04:40

[pdq]

These results suggest that the archiving community should consider using high-sampling conversion to ensure transparency even if the recording is made with standard audio-bandwidth transducers, and when digitizing older recordings made with bandwidth-limited analog systems.

This is equivalent to saying that programs will run better if we insert lots of NOP's. The upsampled data contain the original data plus lots of extra values that are derivable from the original but provide no new information.

[Kees de Visser]

i just was linked this article Which of the two Digital audio systems best Matches the Quality of the Analog System?

Hi boombaard,
some time ago I've started a thread about this paper. Pity if discussions about the same subject become scattered. Care to join the previous thread ?
The paper was written by highly competent and trustworthy people and IMO deserves a serious discussion in the HA forum.

Best regards,

Kees de Visser

PS:

because unless i'm really, really reading this wrong they're saying that if you upsample to the highest SR your sound card can take before playing it back - no matter the quality of the input material - it will be perceived as 'more natural'.

No, it's not about upsampling. The test compares an AD/DA chain to the analog source at two differents sample rates: 44.1 and 352.8 kHz. This is basically different from upsampling 44.1 kHz data that already have passed a low pass filter.

This post has been edited by Kees de Visser: Dec 24 2007, 10:14