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Everything posted by bluesman

  1. But that output amplitude is NOT the same value for every DAC that reads the same 0 dB FS digital signal - there is no specific "equivalent" full scale analog output voltage for a 0 dB FS digital signal because there is no consistent mathematical relationship of any kind between the digital signal being read by the DAC and the analog signal it creates from those digital instructions excepet for the spectral content. Every DAC design outputs the same analog waveform (+/- whatever distortions it intriduces) but at a level determined ONLY by its own design. Whether you think that the
  2. You really don’t understand this. 0dBFS (Full Scale) is the clipping point for a signal in a digital audio product. Rather than measuring from the noise floor up, digital signals are measured (or referenced) from full scale down. A 0dB FS (Full Scale) signal contains the maximum amount of digital information that can be used to represent the signal being defined. The output of a DAC driven with a 0dB FS signal should be at the full electrical potential of the device - this is the voltage drop across the load presented by the next stage’s input. There is no standard for this value
  3. ...which is exactly why AES 17-1998 is confusing many in this thread. Although entitled “measurement of digital audio equipment”, it is based entirely on measuring analog parameters at the ADC input and DAC output. The amplitude of a pure digital signal is determined by its coded content, which is read rather than measured. A 0 dB FS digital signal in a digital domain is defined as the highest level achievable by the EUT. For a 16 bit signal that would be the coded word 32768, which is what constitutes the signal that enters the DAC stage. A steady state sine wave or music waveform at 0dB F
  4. The entire document and its intent are confusing. It refers to an analog input driving a digital device, which is already a bizarre and conflicting circumstance for a document that purports to be for the "measurement of digital audio equipment". And it describes the input signal as "... the maximum analog signal that may be applied to the device for correct operation". Pure digital devices do not have analog inputs, but the document specifies analog signal generators. So there are ADCs and DACs in the mix to provide a digital input and an output signal with measurable voltage. Remember t
  5. We need to be honest and clear when discussing things based on a well defined standard. First, #1 is not your observation - you're quoting definition 3.4 in the very AES standard under discussion. Here's a link to the actual document, so everyone can read it. Second, while it's true that a square wave contains more energy than a sine wave of the same frequency and amplitude, the standard specifically tells us in the ssentence after the one you offered as your observation that only a sine wave is to be used, specifically: "Square-wave signals at this level are not recommended bec
  6. AC born and raised! Back in the '50s and early '60s, the old boardwalk hotels had a lot of dances for kids and parties year 'round. I wasn't in that social circle, but I'd been playing locally at our community centers etc from the time I was about 10 - so I was known to (and apparently liked by) the group that did go to them. Between classmates who went to those events and their parents who knew my family and trusted me, I was able to get regular work playing them with drums, bass and tenor. Then I got my band a regular gig my first year in high school playing for radio station
  7. Absolutely! And I'm nominating you for the 2021 "No Pun Intended" award for "he came out in a wheelchair and blew us all away".
  8. I appreciate it, but I'm 'way too old to try to keep my chops up on wind instruments. After playing piano for 70 years and guitar for 65, I'm finally developing some grace on both. So I've chosen to focus my energy on further development of those skills. 😉 My use of trumpet and sax is probably of interest (and amusement) to you and many AS people, and it's consistent with the thrust of my AS article series on realism vs accuracy in audio systems. I've probably played 2000+ weddings, Bar Mitzvahs, etc and at least as many jazz dates. But I love to play the blues, and I've bee
  9. The concept of worth is purely subjective, Phil. With so many wonderful audio products now available at a fraction of the retail costs of their predecessors, MSRP and market/street price lack strong and consistent correlation with sound quality. And with so many new methods and materials available, even design and build quality are less strongly tied to cost than they used to be. My audio source and close friend for decades was a crusty lawyer-turned-dealer who opened one of the few high end shops in about 1950. One of his many pet peeves was people who said things like "they so
  10. The PL should be an outstanding match for the Fortes. In case you missed it, there's a 2019 Stereophile review of the Forte here that lays it all out well. I assume you're getting the Forte IV, since the III is out of production, and that's reportedly even better than the III. I haven't heard a IV, but the IIIs I've heard sounded quite fine to me. The Klipsch / tube amp combo is truly old school (and an excellent example) - I could live with it very happily, and from your posts I suspect you'll do just that.
  11. Just list them in the classified section on this site - the link is in the header.
  12. Maybe that note was inspired by my favorite piano story. Legend has it that when Rubenstein struck middle C, it sounded better than when anybody else did! PS: I've been a Prima Luna user and lover since soon after Kevin Deal started bringing them in. Your EVO is a wonderful amp that should get the most out of almost any speaker. My KT88 Prologue has been living very happily under our Yamaha grand for almost a decade. Despite its "low" power rating, it's had all the juice needed to drive the many speakers I've had on it, some with sensitivity well below the Tektons
  13. Hi and welcome to the AS community, Phil! I have no experience at all with Tekton or their products, so I can't help you with that. But we see so many rants about similar problems with so many products and vendors that I thought yours called for a bit of clarification. I've been a professional musician for 60+ years (Local 77). Although my main instruments are guitar and keys, I follow at least one web forum for each instrument I've played over the years, of which sax is one. I've never played one of yours myself, but I can tell you (and any on AS who might be tempted to take
  14. If the CD is truly a watermarked disc, it was almost certainly a promo or evaluation copy never intended for distribution beyond the original recipient(s) - and it's probably a CD-R, not a standard CD. This was how demo and advance releases were protected against unauthorized use for about 15 years. Google "Sony rootkit" for info on the biggest debacle of them all in this regard. This one was not for protection of pre-release content. Sony embedded software in all its CDs that wouldn't let you make a copy, and when it was activated it would affect your computer's functionality and alleged
  15. PLease tell us what you think is incorrect. If you're disputing that clipping and headroom manifest entirely differently (and thus need to be treated differently) in analog and digital signals, here's just one of an endless stream of factual support for what I said, which was that "...with digital signals, there is neither headroom nor clipping as we understand them in the analog world". You may have missed the bold, underlined, italicized phrase that ends this sentence. This quote comes from modernmixing.com: "With floating point processing, even though the clipping
  16. With all due respect, I never made such a statement - if you think I did, please quote it so we can all see and understand what you think you saw. A digital signal file is indeed a discrete entity that exists "in isolation", despite your erroneous insistence a few posts above that "[d]igital signals are turned into analogue signals. They don't exist in isolation". SInce you don't seem to understand what a digital signal is, here's one you can see in isolation - it's a snippet of an mp3 (shown in hex format): If that digital signal doesn't "exist in isolat
  17. Of course they do, just as this post exists in isolation. Digital signals are pages of words that can be read, duplicated, edited, printed on paper, archived, etc. They aren’t “turned into” analog signals - they’re used as a template for the creation of de novo analog signals whose vacillations reproduce encoded patterns of change to a level of accuracy determined by sampling rate, bit depth etc. The process of DA conversion is conceptually like the process of amplification. The signal on a record, CD or other source is not turned into the signal that powers your speakers. It’s
  18. And with digital signals, there is neither headroom nor clipping as we understand them in the analog world. Analog devices are generally tolerant of minor level excursions over whatever 0 dB reference is used. Distortion, compression etc are progressive in analog circuitry and increase proportionately (but not linearly) as the level goes further above 0. But when a digital signal level hits 0 dBFS, the maximum # of 1s has been reached for the word size being used and further increases just discard the additional information. There is no soft clipping in digital audio, and there are no inno
  19. I don’t want to continue beating my head against a stone wall here. But the most recent posts have injected a new level of misinformation and confusion that has me screaming inside my head - and I have to relieve the pressure. You can not commingle traditional dB measurements and terms with dBFS (or dB FS, since the omission of the space upsets one participant despite the fact that almost everyone omits it). The “full scale” unit of measurement is ONLY used for digital signal levels in digital equipment. It is a totally different metric from the rest, which are to be used ONLY fo
  20. I let the edit function time out - here’s the rest of the story: The range and distribution of the SPLs in the first part of my response are the acoustic energy analogues of the same mathematical functions derived for line level voltages in the system while listening - so the usual range of listening levels is mirrored by the usual signal levels in the signal path during listening. A distribution function is defined by location, consistency, scale, shape, and skew. Location is determined by measures of central tendency - mean, median and mode. Scale is the range of high
  21. So we now know from the quoted post that you don’t think most audiophiles listen to most music on most systems within a similar range of average SPLs. In my limited experience of about 60 years of active listening in hundreds of homes with friends, fellow musicians, and people trying to sell equipment, I’ve observed the opposite - most listen casually at average (there’s that word again) SPLs somewhere between 70 and 80 dB with a bump to 75-85 when using the music as backing while practicing their instruments etc. There seems to be a somewhat normal distribution (and I hope you
  22. Thanks, but I was being educationally ironic (my just-invented term for using sarcasm as a teaching aid). If I was a bit too sarcastic, thanks to my level of frustration over the refusal to accept that RMS voltage is an averaging measurement, I apologize. The crest factor is yet another example of the same concept applied at a different point in the signal path - it’s probably most useful to recording engineers, mastering labs etc because of the broad range of average-to-peak content in the sounds of individual instruments. It can also serve as a ratio of sustained to transient con
  23. You don’t need to post examples. Of course it’s a generalization, just as “consumer line level” specs are generalizations. And the 0.316 / 2 volt spec pair is a generalized compromise reflecting real world use. But the concept is what’s important in this discussion. Whether you call it usual, general, average, mean or something else, common use of a design spec for a common operating parameter that serves as a proxy for the amplitude of your music at your usual listening level helps to assure compatibility among multiple brands and designs of devices. In this case, we’re discussing the gen
  24. Wow - this is news! So the maximum volume level of the transient peaks in most music is about 5 times the level of the bulk of its content. We should name that ratio because it seems so useful. Let’s figure out a good name together. Maybe there’s a way to measure and specify the two parameters in your 1:5 ratio. The 1 is the general overall level of the main sonic content of music - that sounds kinda sorta like it could be the average level. The 5, which represents the level of the highest peak, is an instantaneous measurement we could call (drum roll please...) the peak. How
  25. I’m sorry to seem contrary, but I don’t know how else to say this - you’re simply wrong. The quadratic mean (also called root mean square and abbreviated as RMS) is another type of average. It measures the absolute magnitude of a set of numbers, not their arithmetic average. So every RMS voltage measurement is an average by definition. Not all means are simple mathematical means. This horse has now been pronounced dead.
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