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Improving loudspeaker crossovers (SBL's)
David Looser wrote: "Eeyore" wrote David Looser wrote: Mind you phase jitter caused by the mechanical parts of a poor quality or faulty CD system *can* stress the error-correction to the point where the playback quality is compromised) NO. Pure horse manure. The digital signal is buffered to buggery. A bit of jitter won't bother it. This is an example of analogue-style thinking being inappropriately attached to digital signal paths. It's not "horse manure". No amount of buffering will cure the problem of recovering data from a badly jittered data-stream. It's the clock recovery that is affected. I can tell that you've never had to design clock-recovery circuits. Actually .... I have been involved in a project to determine weaknesses of early examples of precisely this kind of circuit. There certainly were problems in the early days. If it's *badly* jittered you have a point but I've not seen jitter greater than a couple of ns on digital audio data streams. Graham |
Improving loudspeaker crossovers (SBL's)
"David Looser" wrote in message ... "Keith G" wrote in message ... "David Looser" wrote in message ... Another interesting comment, as it appears to directly contradict your earlier one :-) What comment is that? quote In the case of my son, his kit (Technics/B&W) masks any differences between CD and LP unquote OK, but I would suggest that if you are going to reference previous comments you should include them in your post; I for one will not trawl backwards and forwards trying to work out what you are referring to.... David, all that mastertape horse**** was dealt and dispensed with in here years ago - here's a hint: What "horse****" is that? If you don't *know*, don't guess.... So enlighten me. Difficult to refuse so polite and elegant a request, so here goes: This is an 'audio' newsgroup primarily concerned with the production/reproduction of sound for (mostly) pleasure purposes populated by (a few) people with differing views which I believe can be mostly divided into two main groups - the 'accurists' (for want of a better word) who strive for a sound which is 'as close to the original recorded signal as possible' and the 'realists' (like me) who seek a sound which is 'as close to the original physical sound as possible' or at least their/my *idea* thereof ? (It all becomes highly subjective by the time the recorded sound reaches the listeners ears in his own listening environment when, for example, no two people would agree on the exact same *volume setting* - never mind anything else!) Whatever. One thing the accurists have to fall back on is the 'mastertape' and the gauged faithfulness to this (fidelity) was frequently thrown into the arguments as the be-all and end-all of sound reproduction. The reason for this is that it is fairly easy to *measure* deviation from the original signal, rules can be made from various measurements and it therefore becomes a useful weapon. Where it becomes horse**** in my book is when reference is continually made to mastertapes that no-one has ever heard (or ever will) and to events that weren't witnessed personally. OK? All the realists have to counter this with is that a sound which may not measure too well (and is therefore 'distorted') frequently sounds better (and more *real*) - PW, the accurist's God, said summat like the objective of good hifi' is/would be a 'straight wire with gain'; I say fine, but I'll bend it until it sounds better! Here's an example of how it works which is less than 24 hours old: Yesterday, a very nice blokey came here to hear my Dynaco monos (which are up for sacrifice in the most vicious 'hifi pogrom' yet) and he brought his Bryston preamp, as I don't now have one. This meant I couldn't do anything other than start from stone cold, which I did - the 'sound' was horrible from the off and only started to get better as the Dynies warmed up, but the guy was very curious about the SET amps I have here as he had never heard one! So, after a brief stint when I had swapped his Bryston out and substituted my Denon SS amp (using the pre-outs to drive the Dynacos) which warmed the sound up straight away (distortion) and which the blokey had preferred straight away, I put the Bez Chinese 300B SET on (also up for sacrifice, as it is not of my own making) and instantly the guy sat back and said 'Ah, that's better!' The upshot is my Bezzer is now 8 miles from here, while he checks to see how well it drives his Altec Lansing behemoth speakers (???) - yet another 'realism beats accuracy' triumph, I believe, but I'm not sure about how well it will work on his kit...?? :-) |
Improving loudspeaker crossovers (SBL's)
In article , David Looser
wrote: "Eeyore" wrote in message ... NO. Pure horse manure. The digital signal is buffered to buggery. A bit of jitter won't bother it. This is an example of analogue-style thinking being inappropriately attached to digital signal paths. It's not "horse manure". No amount of buffering will cure the problem of recovering data from a badly jittered data-stream. It's the clock recovery that is affected. I can tell that you've never had to design clock-recovery circuits. Rather depends on your definition of "badly". :-) If you look at the levels of jitter people argue about in audio mags it is of the order of 100's ps to a ns. I'd have said is quite small in the context of SPDIF or EBU, although they are vague about how this is measured. I'd expect this to be easily reclocked and buffered if the engineer knows what he is doing. In practice, my experience is that even a decade old DAC like the Meridian 263 or 563 does this with ease. So does a fairly cheap RX like the one in the Pioneer recorders I have been using recently. Sample-by-sample comparisons using a player with a disc that isn't faulty shown sample-by-sample agreement of recordings. Also agreeing with recordings made directly to computer using a CDROM drive. Can you define what you meant by "badly"? I can see that if the jitter is large enough then you will get problems recovering the actual stream of values. But I can't say I have encountered this with domestic audio kit I've used unless the disc or equipment are faulty. No doubt, though, someone will have made kit bad enough for this to be a problem... :-) Of course, if the 'jitter' has significant amounts at low phase modulation frequencies then the loop or buffer will need to alter the local clock and this will affect the output. I can also see that an RX/DAC that doesn't reclock but simply uses the implict clock in the data will then translate that into phase modulation of the output. But this does not mean that the correct series of values was not RXd. Is this what you were thinking about? That the loop or buffer will respond to close-in phase noise by adjusting the conversion clock frequency? Given values for 'jitter' of the order of 100's of ps I can't immediately see that this would cause a very significant modulation of the DAC clock rate if smoothed with a sensible buffer and loop. Slainte, Jim -- Electronics http://www.st-and.ac.uk/~www_pa/Scot...o/electron.htm Audio Misc http://www.audiomisc.co.uk/index.html Armstrong Audio http://www.audiomisc.co.uk/Armstrong/armstrong.html |
Improving loudspeaker crossovers (SBL's)
"Bob Latham" wrote in message
... In article , Eeyore wrote: However, if those first steps had not been made we wouldn't have the truly excellent medium we have now. It just irks me that the CD standard wasn't say 48kHz, 18 bit. So in your eyes then 16bit 44.1KHz is not transparent otherwise why bother. Cheers, Bob. -- Bob Latham Stourbridge, West Midlands My preference for 48k sampling has nothing to do with audio quality, but to do with compatibility. CD is 44.1, DVD digital TV and digital radio is 48k or multiples thereof. Having CD at 44.1 means that for transmission on TV or digital radio, there has to be an otherwise unnecessary sample rate conversion between source and destination. I know that SRCs are nowadays audibly transparent, but nevertheless, in a professional installation, it adds an unnecesary level of complication. It has caused the BBC for one, considerable extra complication and expense when designing their new Broadcasting House infrastructure, as they decided that network radio would run their studios at 44.1k, whilst News and Current Affairs would run their studios at 48k so as to be compatible with TV. They consquently had to provide both 44.1 and 48k routing, and for DAB and DSAT distribution, they had to SRC the network outputs before transmission. It would have been so much easier if CD had run at 48k. S. -- http://audiopages.googlepages.com |
Improving loudspeaker crossovers (SBL's)
On Sun, 6 Jan 2008 18:08:08 -0000, "Serge Auckland"
wrote: "Bob Latham" wrote in message ... In article , Eeyore wrote: However, if those first steps had not been made we wouldn't have the truly excellent medium we have now. It just irks me that the CD standard wasn't say 48kHz, 18 bit. So in your eyes then 16bit 44.1KHz is not transparent otherwise why bother. Cheers, Bob. -- Bob Latham Stourbridge, West Midlands My preference for 48k sampling has nothing to do with audio quality, but to do with compatibility. CD is 44.1, DVD digital TV and digital radio is 48k or multiples thereof. Having CD at 44.1 means that for transmission on TV or digital radio, there has to be an otherwise unnecessary sample rate conversion between source and destination. I know that SRCs are nowadays audibly transparent, but nevertheless, in a professional installation, it adds an unnecesary level of complication. It has caused the BBC for one, considerable extra complication and expense when designing their new Broadcasting House infrastructure, as they decided that network radio would run their studios at 44.1k, whilst News and Current Affairs would run their studios at 48k so as to be compatible with TV. They consquently had to provide both 44.1 and 48k routing, and for DAB and DSAT distribution, they had to SRC the network outputs before transmission. It would have been so much easier if CD had run at 48k. S. But their analogue terrestrial service needs NICAM at 32k. Yet another sample rate conversion. Conversion is such a trivial matter these days that I would suggest that it need not be considered in making technical decisions. d -- Pearce Consulting http://www.pearce.uk.com |
Improving loudspeaker crossovers (SBL's)
Keith G" wrote in message
... This is an 'audio' newsgroup primarily concerned with the production/reproduction of sound for (mostly) pleasure purposes populated by (a few) people with differing views which I believe can be mostly divided into two main groups - the 'accurists' (for want of a better word) who strive for a sound which is 'as close to the original recorded signal as possible' and the 'realists' (like me) who seek a sound which is 'as close to the original physical sound as possible' or at least their/my *idea* thereof ? (It all becomes highly subjective by the time the recorded sound reaches the listeners ears in his own listening environment when, for example, no two people would agree on the exact same *volume setting* - never mind anything else!) In the world of cinema sound the cinema's playback system is conceptually divided into two parts at the source selector switch: the A-chain, which is the soundhead, pre-amps decoders etc, and the B-chain, which is the balance of the system, volume control, equalisers, power amps, speakers plus, of course, the acoustic environment. There is a separate A-chain for each type of source: optical analogue, magnetic analogue, optical digital, hard-drive etc. whilst the B-chain is the same for them all. Films are mixed in dubbing theatres which conform to an international standard for frequency response, reverb time etc. and cinema B-chains are adjusted to conform as closely as possible to that standard. Meanwhile the A-chain is adjusted to be as nearly transparent as possible, ideally the electrical signal leaving the cinema A-chain is exactly the same as the electrical signal leaving the sound mixer's desk. This ensures that the sound heard by the cinema patron is as close as possible to that heard in the dubbing theatre. It also means that, at least as far as film sound sources are concerned, the volume and frequency response are similar regardless of the type of soundtrack being played (there is remarkably little audible difference when switching back and forth between the digital and analogue soundtracks on modern 35mm film prints) Transferring this concept to the home Hi-Fi area, the record-player plus RIAA pre-amp or CD player, tape machine, tuner etc. is clearly the A-chain. Whilst the volume and tone controls, power amps and speakers (+room of course) are the B-chain. Now I grant you that there is less standardisation of music mixing rooms than is the case with film dubbing theatres, and there is no standardisation of home listening rooms at all, nevertheless it seems to me that at least for the A-chain the same principle applies, the signal leaving the RIAA amp or CD player should be as close as possible to that entering the monitoring chain in the mixing room. If you want to bugger-about with the sound to make it "better" you can do that with the B-chain. Whatever. One thing the accurists have to fall back on is the 'mastertape' and the gauged faithfulness to this (fidelity) was frequently thrown into the arguments as the be-all and end-all of sound reproduction. The reason for this is that it is fairly easy to *measure* deviation from the original signal, rules can be made from various measurements and it therefore becomes a useful weapon. Well no. The reason is that the mastertape represents the sound that the sound engineer, record producer, artist etc. agreed was what they wanted the record to sound like. Where it becomes horse**** in my book is when reference is continually made to mastertapes that no-one has ever heard (or ever will) No-one? what about the recording engineer, the producer, the artist etc? and to events that weren't witnessed personally. OK? All the realists have to counter this with is that a sound which may not measure too well (and is therefore 'distorted') frequently sounds better (and more *real*) Well yes I understand, That's why 1950s AM radios with single-ended output pentodes running 10% distortion at 3W sound far more "real" than anything else! :-) - PW, the accurist's God, said summat like the objective of good hifi' is/would be a 'straight wire with gain'; I say fine, but I'll bend it until it sounds better! Here's an example of how it works which is less than 24 hours old: Long rambling story snipped. The upshot is my Bezzer is now 8 miles from here, while he checks to see how well it drives his Altec Lansing behemoth speakers (???) - yet another 'realism beats accuracy' triumph, I believe, How do you work that out?, what did your story have to do with either accuracy or realism? David. |
Improving loudspeaker crossovers (SBL's)
"Don Pearce" wrote in message
... On Sun, 6 Jan 2008 18:08:08 -0000, "Serge Auckland" wrote: "Bob Latham" wrote in message ... In article , Eeyore wrote: However, if those first steps had not been made we wouldn't have the truly excellent medium we have now. It just irks me that the CD standard wasn't say 48kHz, 18 bit. So in your eyes then 16bit 44.1KHz is not transparent otherwise why bother. Cheers, Bob. -- Bob Latham Stourbridge, West Midlands My preference for 48k sampling has nothing to do with audio quality, but to do with compatibility. CD is 44.1, DVD digital TV and digital radio is 48k or multiples thereof. Having CD at 44.1 means that for transmission on TV or digital radio, there has to be an otherwise unnecessary sample rate conversion between source and destination. I know that SRCs are nowadays audibly transparent, but nevertheless, in a professional installation, it adds an unnecesary level of complication. It has caused the BBC for one, considerable extra complication and expense when designing their new Broadcasting House infrastructure, as they decided that network radio would run their studios at 44.1k, whilst News and Current Affairs would run their studios at 48k so as to be compatible with TV. They consquently had to provide both 44.1 and 48k routing, and for DAB and DSAT distribution, they had to SRC the network outputs before transmission. It would have been so much easier if CD had run at 48k. S. But their analogue terrestrial service needs NICAM at 32k. Yet another sample rate conversion. Conversion is such a trivial matter these days that I would suggest that it need not be considered in making technical decisions. d -- Pearce Consulting http://www.pearce.uk.com That's what I would have thought, especially considering the BBC desks have SRC at each input, but they, in their infinite wisdom, decided that they will run studios at the sample rate of their primary input source (CDs in the case of radio, VTRs in the case of TV) and sample rate convert for each destination. S. -- http://audiopages.googlepages.com |
Improving loudspeaker crossovers (SBL's)
"David Looser" wrote in message ... Keith G" wrote Whatever. One thing the accurists have to fall back on is the 'mastertape' and the gauged faithfulness to this (fidelity) was frequently thrown into the arguments as the be-all and end-all of sound reproduction. The reason for this is that it is fairly easy to *measure* deviation from the original signal, rules can be made from various measurements and it therefore becomes a useful weapon. Well no. The reason is that the mastertape represents the sound that the sound engineer, record producer, artist etc. agreed was what they wanted the record to sound like. Irrelevant - whatever the mastertape is (or isn't) is not the issue; the 'accuracy vs. realism' argument is.... Where it becomes horse**** in my book is when reference is continually made to mastertapes that no-one has ever heard (or ever will) No-one? what about the recording engineer, the producer, the artist etc? More irrelevancy - they weren't present in the arguments in this group, AFAIA.... Long rambling story snipped. Nothing like as long as your 'cinema irrelevancies' which I snipped... The upshot is my Bezzer is now 8 miles from here, while he checks to see how well it drives his Altec Lansing behemoth speakers (???) - yet another 'realism beats accuracy' triumph, I believe, How do you work that out?, what did your story have to do with either accuracy or realism? Because the guy leapt at a 300B SET (over his own SS pre and PP valve monos) which, as I'm sure you are aware, are slated endlessly in this group for 'distortion' ('broken' even...) but which I maintain provide a more *realistic* sound - kinda proves my point, doesn't it? Asitappens, he's been on the phone since to say (verbatim) 'I'm not getting the same, sweet sound as you get on your Lowthers!' - I'm not surprised, I didn't think he would but I was willing to let him try! While I'm on: I have started a ferocious campaign to reduce clutter and am punting no end of stuff out on eBay - I have 5 auctions on now, including a set of both Morgan Jones 'Valve Amp' books which may be of interest to someone he http://cgi.ebay.co.uk/ws/eBayISAPI.d...MESE:IT&ih=017 Condition *as new*.... (Sorry to say! :-) |
Improving loudspeaker crossovers (SBL's)
"Keith G" wrote Correction: Because the guy leapt at a 300B SET (over his own SS pre and PP valve monos) The (Bryston) SS pre was his; the valve monos are mine... |
Improving loudspeaker crossovers (SBL's)
"Jim Lesurf" wrote in message
... Can you define what you meant by "badly"? I can see that if the jitter is large enough then you will get problems recovering the actual stream of values. But I can't say I have encountered this with domestic audio kit I've used unless the disc or equipment are faulty. No doubt, though, someone will have made kit bad enough for this to be a problem... :-) I've no experience in this respect with domestic kit, or with CD as source, so I will accept the consensus here that with CD this isn't a problem. My experience is with rotating-head tape transports and with long (many km) transmission lines running at bit rates of several hundred Mb/sec. (Yes transmission lines can cause jitter, as well as often having a very poor eye.) David. |
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