Monday, June 25 11:00 – 12:30Paper 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 Reefman31McGill University, Montreal, Quebec, Canada2Centre for Quantitative Methods CQM BV3Philips Research, Eindhoven, The NetherlandsIn 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.
In any case it appears to be a test to see which of two digital systems sounds most like an analog one, which is considered 'technically best'. I'd be curious to see the reasoning behind such a claim. Are papers being presented, peer-reviewed?
I will end this month's essay by quoting, from a paper given at the conference, the results of experiments on the audibility of high sampling rates: "To achieve a higher degree of fidelity to the live analog reference, we need to convert audio using a high sampling rate even when we do not use microphones and loudspeakers having bandwidth extended far beyond 20kHz. Listeners judge high sampling conversion as sounding more like the analog reference when listening to standard audio bandwidth." (footnote 2)So that's that, then.
So far though I haven't been able to find any details of the presentation. From the abstract , if nothing else it looks to me like M&M's sample pool was much deeper....
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 100kHz microphones and added super-tweeters.
(1) Is there a possible benefit of high-resolution audio at lower frequencies without the necessity for the reproduction of supersonic components?
Is a noticeable benefit of high-resolution revealed in surround sound listening?
This means that at the 100 kHz bandwidth the low sampling system Y is more often than X judged to be most like the analog system A.
This means that with the cut-off at 20 kHz bandwidth the high sampling system X is more often than Y judged to be most like the analog system A.
In the conclusions, they hypothesize that the wide bandwidth system sounds less transparent because of what they call "noise-like artifacts" in the high frequencies (above 20kHz), which are not passed in the lower sampling rate version of the wide bandwidth system. I am not sure how they support this hypothesis, given that the analog reference system described also passed the ultrasonics.
It's also important to know that only two sample rates were used: 44.1 and 352.8 kHz, derived from a 128xFs 5 bit delta-sigma modulator. It would have been interesting to know if popular standards like 96 and 192 kHz would behave differently. I can imagine however that time and budget were limited.
Note also that only one brand of AD/DA converter has been used (Digital Audio Denmark). AFAIK it hasn't been tested if results remain identical with other brands.
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.
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.
PS:Quotebecause 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.
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'.
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.
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?
Introduction:The audio archiving community responsible for the preservation of our sonic cultural heritage is interested in adopting a digital conversion and storage format that can be considered transparent by listeners skilled in the art of audio. Therefore, a digital medium having high degree of fidelity to the analog reference is needed.
The whole thing about the lowpassed high resolution version being perceived as better than the non-lowpassed high resolution is very interesting.
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'.
The paper gives some evidence that high sample-rate formats provide a copy that is sonically closer to the original compared to standard rate formats.
This one is not flawed, it just failed.
Assumption 2: “Both digital systems X and Y have aperformance that is not better than the performance of theanalog system A, if we measure this performance on theone-dimensional latent variable.”Assumption 2 has reasonable logic from audio knowledgeand experience if we assume that in theory the analogversion has an infinite number of data points (that is,infinite resolution), and DXD (8 Fs) has used eight timesmore data points for conversion than the 1-Fs system.
Considering that the experience of professional recordingand mastering engineers working with high-resolutionaudio has not been confirmed and quantified in laboratorytests, we do not have a good validation of the testingmethods used in our industry. The current subjective testingmethodology does not seem to sufficiently reveal oramplify the features characterizing individual listeningexperiences. Perhaps this methodology, which is derivedfrom food and fragrance testing, is not as readily effectivefor investigating subtle auditory sensations and the experienceof music? We too would like to encourage morework in this area.