There is very little conversation about time variant behaviours, which is a shame. IME, this is one the biggies - a system will often sound different half an hour later from what it does currently, and this has absolutely nothing to do with acclimatising to the sound! Sometimes it will sound better, sometimes worse, and the speed at which it changes will vary, for just about every reason one can think of. So, something like ABX testing can give meaningless results for that reason alone.
One of the most extreme examples of that type of "transitioning" I've heard occurred on the ambitious DEQX system of the member up the road. Initially it sounded flat, lacking in dynamics
, for about an hour - then, in the middle of a song, over a period of a minute of two the sound came together, just like that - it became "musical". He picked it too, of course - and it was because the DEQX unit had been powered down for some time beforehand, because of an earlier electrical storm threat - something in the unit needed of the order of a couple of hours conditioning to snap right, which I'm positive no conventional measuring would pick up ...
One of the most extreme examples of that type of "transitioning" I've heard occurred on the ambitious DEQX system of the member up the road. Initially it sounded flat, lacking in dynamics
, for about an hour - then, in the middle of a song, over a period of a minute of two the sound came together, just like that - it became "musical". He picked it too, of course - and it was because the DEQX unit had been powered down for some time beforehand, because of an earlier electrical storm threat - something in the unit needed of the order of a couple of hours conditioning to snap right, which I'm positive no conventional measuring would pick up ...
No one has demonstrated there's a need on the playback side. It's just been a lot of hand waving.
se
not true... gave sources already of F.Toole and dblt of 16bit. many others now.
What hasnt been shown a need for (yet?) is 32bit playback.
THx-RNmarsh
The "standard" you keep presenting is THD + noise as an rms level over a BW. SFDR relates to communications more than audio i.e. radios cell phones.
There is very little conversation about time variant behaviours, which is a shame. IME, this is one the biggies - a system will often sound different half an hour later from what it does currently, and this has absolutely nothing to do with acclimatising to the sound! Sometimes it will sound better, sometimes worse, and the speed at which it changes will vary, for just about every reason one can think of. So, something like ABX testing can give meaningless results for that reason alone.
One of the most extreme examples of that type of "transitioning" I've heard occurred on the ambitious DEQX system of the member up the road. Initially it sounded flat, lacking in dynamics
I am quite sure conventional testing would have picked it up. Completely and totally.
Oh Please. it was an illustration of what peak levels were being refered to regarding dynamic range.
IMO --- We need to measure the actual dynamic range of the ADC or DAC completed as a product for sale. Forget the what the theory says for the moment and see what the measured data says about what we actually have got in our hands. The measurement is clear yet many keep talking like they can just make up what they want for a number based on not much of anything. lets measure it correctly (eg by the standards), normalize the data and that means with all harmonics and extraneous stuff. I would use peak levels.
When this is done, the literature indicates much higher than detectable thresholds for many products.... esp 16bit. Thus, the 24 bit and higher we see today- Its all practically just history by now. The industry has moved on to 24 bit playback with research behind the shift. I've only touched on a point here and there to make this story as short as possible.... 24bit now.
What is the real question? That 16bits is all we need? If so, they are far behind and need to catch up.
THx-RNMarsh
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The reality is that digital playback very often fails to present dynamic range well - and this is a failing of the overall system, at that point in time. It's a form of distortion, just like pops and crackles and rumble are in vinyl - and can be fixed by sorting out the underlying issues, or various workarounds. I have been amazed at times, listening to CDs I know well on other systems, where half the sound just completely disappears - the low level information is totally discarded, some electronic "black hole" has completely swallowed it up ... .
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I have the PRO version (not sold to consumers) -- an approved certified installer only item. Havent used it enough to say it does or doesnt do that but that i havent heard it.
THx-RNMarsh
I would use the peak value as seen on an FFT or spectrum analyzer to establish min floor range. But, that may be too harsh for little ol cheap consumer gear. I would also add many more tones to simulate and stimulate max 'spurs' if going for averaging. it doesnt seem reasonable to me to not stimulate to the fullest extent possible how the equipment is actually going to be subjected..... a single sine wave tone is very far from how it will be used. yet, we are to draw some audible conclusions from that number?
But that is just me.... I want to know afterwards if those numbers from more heavily exercised tests are closer or past the audible thresholds people claim to hear.
Anyway, it started for me as 16 isnt good enough.... 24 is a lot better from what I can hear and why -- more bits, fewer processes from pre/post processing and spinning discs(numbers and listening).... would 32 be better yet for some yet to be known reason.... perhaps lower distortion at the mid to low level - where we spend a lot of time listening.
There must be an equivalent to the ADC tests for DAC's.... But, I found this 'guidelines' for testing ADC which is very illuminating on the need to test further than mfr specs.
www.digitizationguidelines.gov/audio-visual/documents/ADC_Perf_Test_2012-02-24.pdf
THx-RNMarsh
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It's always going to be easy to "prove" that 24 bits sounds "better" than 16 ... simply by having the playback mechanism that's decoding the 24 bit recording doing a better job than the one dealing with the 16 bit version. One reasonable approach is to truncate "proper" 24 bit recordings to 16 bits within, still being 24 bits on the outside, and compare on the same replay chain - doing any thing less scrupulous than that has zero value as an experiment .
PS Audio's DirectStream sounds very interesting, converting digital to "pure" 1-bit DSD - effective resolution apparently about 17 bits - comments indicate that the implementation has been done well, bypasses some of the typical "digital sound" gremlins ...
Edit: I note particularly that this review, http://www.6moons.com/audioreviews2/psaudio/1.html, which I just read, nails what it's doing well.
Edit: I note particularly that this review, http://www.6moons.com/audioreviews2/psaudio/1.html, which I just read, nails what it's doing well.
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Interested in your opinion it uses the same ESS DAC as the Chinese are putting in their next gen cell phones. As far as I can see the only difference is the discrete I/V. The Chinese are using SOTA op-amps BTW.
Missed this from you ----- first impressions are very positive (compared to various iPOD). I have not compared to the Benchmark-2, yet. If we continue in this direction with mobile and clouds and the like, the sound is going to get better for everyone, over-all. And with sota opamps, too. Who would have thunk it possible? It is a big step forward that we can all take advantage of... if we choose to.... its there/here. Now we are getting somewhere.... finally! Now that my hearing isnt as perfect as it used to be :-( Better late than never.
THx-RNMarsh
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