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AES conference London: High Resolution perception, paper about listening test
Kees de Visser
post Jul 5 2007, 11:25
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Unfortunately I wasn't able to visit the june 2007 AES Conference in London about High Resolution Audio.
The paper/presentation about a high-res audio listening test seems interesting. I'm wondering if anyone on HA happens to have been there and can share some information.

preview of the paper session:
QUOTE
Monday, June 25 11:00 – 12:30
Paper Session 2 — Perception

2-1 Which of the Two Digital Audio Systems Meets Best with the Analog System?— Wieslaw Woszczyk,1 Jan Engel,2 John Usher,1 Ronald Aarts,3 Derk Reefman3
1McGill University, Montreal, Quebec, Canada
2Centre for Quantitative Methods CQM BV
3Philips Research, Eindhoven, The Netherlands

In this listening test, two digital audio systems (B and C), and one analog system (A) were tested by 10 test persons who listened to a surround sound scene “live” (without recording). The main question to be answered was: “Which of the two digital systems meets best with the analog system?” Both digital versions had 24-bit dynamic resolution but differed in sampling rate with which the analog signal was sampled. One version © was sampled with a CD rate of 44.1 kHz, the other (B) 8 times faster. There were also two test conditions, where in one condition there was a bandwidth cut off at 20 kHz instead of the 100 kHz that was possible with special 100 kHz microphones and added super-tweeters. For each subject, the experiment was replicated six times, in each of the two conditions. The outcome of each experiment was a 0 or 1, where the 1 means that the, technically best, digital system B has been chosen as meeting the analog quality. The paper describes the test and the outcome.

Without having read the paper, it's not clear to me whether the test was double-blind or not. Apparently it was not possible to replay sources, since the audio source was "live". How reliable would a test like this be ?
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boombaard
post Dec 24 2007, 12:02
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must've missed it when you posted it first.. odd.

anyway

QUOTE
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:
QUOTE
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).

QUOTE (Kees de Visser @ Dec 24 2007, 09:04) *
PS:
QUOTE
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.


well, the topic of the paper is at least remarkably on-topic.. but i don't really see what you base your statement about the AD/DA chain on.
as i understand it, both the 20kHz recording equip and the 100kHz ones were attached to the same chain, so that doesn't really seem relevant here.
furthermore:
QUOTE
In the 20 kHz case (C2), they liked the high-sampling version, which would have an effect on other aspects of technical quality than the high frequency response. This may indicate that we do not need 100 kHz microphones and 100 kHz loudspeakers, but we do need 100 kHz capable recorders.


so what are they saying here then? that 20kHz cutoff mikes still produce overtones/noise in the 20-100kHz band that's relevant to the recording and is perceived as adding to the perceived 'naturalness' of said recording, even though they're not 'real'?

anyway, what i don't really see is why that would influence the archiving community.. i can see how this might be interesting to recording studios, but by the time it reaches the consumer, the format is already decided on. so why would it be relevant to him to save it in as high a sampling rate as possible?
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