I could tell you precisely what makes a girl beautiful to me. Similarly, I can describe precisely what make music enjoyable for me, so that I can actually consciously pay attention to those characteristics when choosing equipment or doing tweaks.
Hence you and I are very different, and we're fine with that.
Yeah, but what is 'the overall effect'? What does 'getting transported by the performance' means precisely? I get told by people my rig provides that, these very same words. But I know why, because I know what I was listening for when I tweaked.
If 'frequency response don't enter the picture' what are you paying attention to? Or are you not paying attention? What if others are paying attention and listening for very precise things and finding differences in filters where you simply aren't paying attention or are insensitive to this (totally fine with that too)?
Does your experience negate theirs? Do theirs negate yours?
I think neither is the case, but if you don't hear something or cannot verbalise emotions or SQ, it doesn't mean other people don't or can't.
Gents, can you please beat up your wives, kids and pets or find other ways outside this thread to release your anger?
But I'd not ask.
Yes, I had already figured that out.
I figured that from what you'd written previously, you understood this already. You wrote :
(I find the clicks and pops annoying - they bring me out of the music, making my enjoyment less: when they occur, they make me think of the quality of the playback chain rather than focusing solely on the musical performance).
When I write 'transported by the music' I think (though I have no evidence to back this up) that its the opposite of what you meant when you wrote 'bring me out of the music'. Prior to being 'brought out of the music' by those clicks and pops you were being 'transported by the music', no?
Already answered.
Certainly possible, equally possible is they're hearing placebo effects.
Why would experiences negate each other? I can't think of a reason myself so these look like irrelevant questions.
I am not trolling. I am asking a simple question and failing to get a straight answer on why oversampling before sending to the DAC makes any change to its 'workload'. If that is trolling sobeit. But I'm out of this thread and back to being satisfied with my vinyl and my digital.
Yes. At some point one must simply abandon conversations which are going nowhere fast. There are a number of people who, for whatever reason, seem to think that a) some ones and zeroes are more "oneish" or "zeroish" than other ones and zeroes, and/or b) you can get something from nothing.
There is no way to convince these people otherwise, so it is time to simply stop participating.
Admittedly I haven't been following the history of this little 'spat' but my post no. 359 provides an outline for why there may be less processing going on within the DAC when the input datarate is higher. The digital filters in S-D DACs are implemented as multirate filters, normally doing interpolation in steps of 2X. If you provide (say) 4X rate as an input, the 2X and 4X rate output filters presumably are disabled? Even if not powered down, they'll be (again presumably) provided with zeros and so generate fewer switching transitions than when fed audio.
Exactly its all presumption. Just trying to find if there is an evidence of this from the proponent.
Clearly this only holds for audio DACs. Those who roll their own from parts and can build their own interpolation stages have more freedom. Hopefully someone who knows will chime in as the ESS data sheets are high on hype and low on useful from my searches. Just looked at the spec sheet for that PCM1716 in my beresford. That internally runs 96KHz with 8x oversampling of the filter. There are 2 digital filter options you can set externally. what they don't explain is how they get to 96KHz in the first place. Suspect the interface chip is expected to do that.
But it does the job for me for now. There is a drop in wolfson replacement part. If one considers QFP drop in!
Clearly this only holds for audio DACs. Those who roll their own from parts and can build their own interpolation stages have more freedom. Hopefully someone who knows will chime in as the ESS data sheets are high on hype and low on useful from my searches. Just looked at the spec sheet for that PCM1716 in my beresford. That internally runs 96KHz with 8x oversampling of the filter. There are 2 digital filter options you can set externally. what they don't explain is how they get to 96KHz in the first place. Suspect the interface chip is expected to do that.
But it does the job for me for now. There is a drop in wolfson replacement part. If one considers QFP drop in!
Thx for clarifying this.
Actually, you can get (rough) information about the 1704 performance at -60db from the graph comparing 16 vs 24bits operation. From this graph, apparently based on a pcm1704k, it appears that the pcm1704 @24/96 is a bit better than a pcm1702k @20/44.1 at -0db but performs the same at -20db and is worse at -60db.
I think however that we are missing data to attribute with certainty that declining performance to the increased sample rate. Since I'm not about to invest a lot of money in order to test this and since I cannot find old posts with relevant measurements, I'll leave it at that.
Browsing datasheets from the various r2r DAC, there's never anything about thd vs various sample rates, while it's quite standard for DS DAC... frustrating.
The dynamic range figure is a way to compare at -60dB - however the 1702 does not inform us whether its 'typical' figure is for a -K or a lower grade. I rather suspect not a -K.
I agree with you its not certain but in the absence of any other hypothesis I'll take the 'its the increased sample rate' one. I've never seen a DS for a DAC where the figures improve at higher rates - if you'd like to see more quantitative data I suggest looking at ADI's TXDACs which, while not audio parts, do demonstrate very clearly how ENOB declines with sample rate.
Bit of a shame ... YashN and abraxalito are actually on the same page, but the language being used is not helping them to understand that - I'm quite sure that if they were listening together to different systems, with different qualities, they would basically agree on what was "good" and what was "bad" - just the description of what they were hearing would differ ... 🙁
Depends which world you're talking about, digital or analogue. Obviously nonsense if talking digital, terribly important if talking analogue. The mental glitch a lot of people seem to have is that if a component or pin is marked "digital" in some part of the circuit that it exists in a completely different 'space' from the analogue bits. Well, guess what, boys and girls, there isn't foot thick shielding and perfect isolation between those lovely theoretical areas, they happen to be mighty close together in electrical terms, in the real world. Yes, "perfect" engineering would mean there is zero interaction - and wouldn't it be marvellous if all engineering was always perfect, 😉 - but is it "perfect" in this little bit of engineering we worry about ... ? 🙂
I never bury myself in the technical whys and wherefores, as is being done here, because my experiences have demonstrated, to me, that these aren't essential to the goal. High quality sound may be achieved by worrying about the n'th degree of technical sophistication, getting one's hands filthy with grease by diving into the heart of the engine; but it can also be achieved by working on the outside of all this, standing well back from the nuts and bolts of the mechanisms.
The best solution, that is, the one that is achieved most easily, with least cost and effort, most likely will occur combining the two approaches - getting the "heart of the machine" to work the best way possible and making sure the "machine" is left alone to do its job with minimal interaction with anything else.
The best solution, that is, the one that is achieved most easily, with least cost and effort, most likely will occur combining the two approaches - getting the "heart of the machine" to work the best way possible and making sure the "machine" is left alone to do its job with minimal interaction with anything else.
Hi Frank thanks a lot for the very valuable advice.
Your words confirms my feelings but also listening impression.
Sometimes i read that high resolution is key for a good digital sound.
Not true. I had the opportunity to listen to a very old Grundig cd 7500 cd player at a vintage audio fair.
https://www.youtube.com/watch?v=N428fRQ8tWg
I understand that it does not reach even 14 bits of resolution. Still i heard an amazing rendering of a piano ... very very good.
It was just 250 USD ... but without remote control.
When i decided to buy it it was already gone.
This to say that digital is tricky. It is not straightforward.
Maybe as someone says it is a matter of getting the right amount and type of distortion we like ? 🙄
Uhm ... dont think so. There is something going on.
Anyway i feel to be close to my final goal that is decent sound out of a computer, any computer that works.
I am focusing usb to spdif converters these days.
I am stuck with AES/EBU connection that to me sounds more relaxed while keeping all the detail.
These converters are very handy because they allow to use any dac around with the correct connections. And dacs are everywhere these days.
In my experience a decent dac when fed with a good digital signal sounds at least decent. Satisfactory.
Thanks a lot again.
Have a nice day, gino
My very first CD player way back in 1984, the Grundig shown incidentally is a rebranded first generation Philips player. (Mine was a Philips Magnavox) Stock it was terrible sounding, replacing certain passives made a pretty big difference, in particular the deglitching caps on the dacs, which were 4x oversampling 14 bit converters... Mine died after about 14yrs..
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I don't think I ever heard my old CD players properly because the quality of CD recordings back then were simply awful compared with what I would now deem to be a good recording.
My approach is completely different - mentally, I "insist" that the CD recordings are all good, and if they don't come up to scratch, subjectively, then it's because my playback system isn't good enough. This forces me to deal with the real underlying issues of poor digital playback - and when this is done, lo and behold, those "bad" recordings emerge fresh and sparkling, full of life and capable of being immensely satisfying, 🙂.
I have all of my early CDs to this day, and TBH, they are far better than the early CD players I played them on - far better in fact than I ever suspected at the time.
I believe a good many of the issues with early digital was the player and not necessarily the recording.
I believe a good many of the issues with early digital was the player and not necessarily the recording.
Iphone 5s and tube gain of 3. nothing beats it with lossless files.
tube sweetens sound somehow
dac does 1v into 330 ohms 0.001 distortion jitter noise -100db
clock in chip locks it in
tube sweetens sound somehow
dac does 1v into 330 ohms 0.001 distortion jitter noise -100db
clock in chip locks it in
Hi and thanks for the kind info.
Actually i do not know if the 7500 i listened to was stock or modded.
But i remember a captivating sound ... musical and robust.
The lack of remote and the fear of transport breaking-down stopped me by buying. Actually i was going back to buy it anyway but it was already sold.
Today i am stuck with the pc that i am trying to make sound good.
Not that easy but so convenient. I have most of my cds on a NAS.
This solution is superb for flexibility. But even only a portable cd can store more than 1400 cds.
Thanks a lot again. Kind regards, gino
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