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Adding to the above, a good tube/valve amp needs more power than the solid state types, so extra accomodation is required there. Sometimes the sound makes it very beneficial.
The sampling theory formulated by Claude Shannon in the late 1940s had a key requirement: The signal to be sampled must be band-limitedthat is, it must have an absolute upper-frequency limit. With that single constraint, Shannon's work yields a remarkable result: If you sample at twice that ratetwo samples per period for the highest frequency the signal containsyou can reproduce that signal perfectly. Perfectly. That result set the foundation for digital audio, right up to the present. Cue the music.
However, in the 69 years since Shannon published "Communication in the Presence of Noise" in the Proceedings of the Institute of Radio Engineers (footnote 1), sampling theory has moved on. The new work began almost immediately, carried out by mathematicians and math-fluent engineers; indeed, some of it had already been done when that paper was first published, in January 1949. But even as Shannon's work was embraced by the digital audio communityin audio, Shannon sampling theory is the foundation for almost everything digitalthat post-Shannon work remained hidden. As recently as the 1990s, when post-Shannon sampling was applied to digital signal processing, the focus was almost entirely on visual informationimaging. With just a handful of exceptions, the audio world was oblivious.
Post-Shannon sampling theory relaxes Shannon's requirement that a signal to be sampledeg, a recording of musicbe band-limited to half the sample rate. Relaxing that constraint restores the symmetry between the time and frequency domains that was missing from Shannon's theory. In the newer theory, it's fine to use an antialiasing filter, but it's not required. Post-Shannon sampling accepts aliasing as a matter of course while allowing its impact to be minimized in both the time and frequency domains.
Some years ago, Bob Stuart and Peter Craventhe creators of Master Quality Authenticated, or MQAbegan exploring some similar ideas in an audio context. Their first article hinting at the technology that eventually became MQA (footnote 2) referred to some post-Shannon work, and in some later writingsincluding "MQA: Questions and Answers," published on the Stereophile website in August 2016the references are fairly explicit. I began to wonder: Is MQA a rigorous application of post-Shannon sampling theory?
Considering the newer theory's relaxation of Shannon's absolute prohibition against aliasing, this seemed a reasonable entry point for a somewhat technical interview with MQA's Bob Stuart on aliasing and its effects in MQA. The interview, carried out mainly by e-mail, is presented here in slightly compressed, lightly edited, occasionally annotated form.
Jim Austin: Is MQA an application of post-Shannon sampling theory to audio coding?
Bob Stuart: Yes, very definitely, it is!
Austin: Is it the first such application?
Stuart: Yes. So far as we know, this is the first.
Austin: Is it a rigorous application of post-Shannon sampling theory?
Stuart: Yes, we believe so. MQA stands on a firm basis which synthesized intuition of desirable characteristics, the mathematics of sampling and reconstruction based on B-splines, losslessly reversible processing (used in flattening)and was informed by empirical observations, auditory modeling, and hundreds of experiments.
Austin: MQA's critics have often focused on aliasing. In a patent application covering MQA technology, you claimed the invention of "a system . . . wherein . . . the asymmetric component of response of the decimation filter is characterized by an attenuation of at least 32dB at frequencies that would alias to the range 07kHz on decimation [footnote 3]. Is that specificationattenuated by at least 32dB at frequencies that would alias to the range 07kHz on decimationrealized in MQA's implementation?
Stuart: In all cases the aliasing heard by a listener with an MQA decoder will be well below that implied in the quoted patent claim and will be, we claim, either inaudible or nonexistent.
[To test this claim, beginning with a FLAC file containing white noise at 10dBFS peak, I pasted in 20 seconds from Talking Heads' "Girlfriend Is Better" at the 20s mark, 50dB below the noise, repeating the music and increasing the level by 10dB every 20 seconds. I can detect very faint drums at the one-minute mark, 30dB below the noise level and at least 30dB louder than the aliased-content level allowed by the MQA specification. By 20dB below the noise40dB above the specI could hear the music clearly. My conclusion: The specified level of aliasing is not audible, with a safe safety margin. The use of white noise instead of a 1/f, music-like signal makes this a very conservative test.
Your mileage may vary, so do the test yourself with the embedded audio file. At first you will just hear the noise, but you will then start to hear the music beneath the level of the noise.]
Footnote 2: J. Robert Stuart and Peter Craven, "A Hierarchical Approach to Archiving and Distribution," AES Paper 9178 (8 October 2014).
Footnote 3: Note that the claim is that aliased content will be attenuated by 32dB, not that it will be 32dB below the regular audio in the specified range. At CD sampling rates, at which aliasing would typically be strongest, aliasing at the top of this range will be reflected down from about 36kHz. In that frequency range, the music in an audio file is already very low in level. I estimate that if this specification is met, the aliased content will be at least 60dB below the musical information at 7kHz.Jim Austin
Adding to the above, a good tube/valve amp needs more power than the solid state types, so extra accomodation is required there. Sometimes the sound makes it very beneficial.
You can use external power conditioners for cleaning up electricity ........ AQ makes several of these power conditioners ........ Also PS audio makes them ....... At least dozen other companies make these power conditioners .......... Many of these including AQ and PS audio are very favorably reviewed in Stereophile ....... RH-5 can put out a lot of "juice" ....... RH-5 has 3 different gain settings ......... It can drive any 'phone ....... I use Hugo2 which is battery powered (internal supply) ........ It can easily drive a lot of high efficiency low impedance 'phones including my Lcd-x and Lcd-MX4 ....... It has problem driving low efficiency high impedance 'phones like my Lcd-4 ...... Hugo2 sounds great with my high efficiency low impedance 'phones ....... Hugo2 is very favorably reviewed by many (and many "golden ear") audio reviewers ........
I like AudioQuest products a lot, so I wouldn't hesitate to buy their power solution for a small amp that puts out maybe 5-10 watts of power. If you do have the power, chances are you'll like the lower efficiency headphones best. A lot of people read headphone specs and get "just enough" power, then they find out that a lot of the amps do a kind of "soft" clipping where the sound isn't too harsh, but some of the dynamics get squished.
Agreed ......... RH-5 puts out enough power, it can drive any 'phone ......... Like I mentioned, it has 3 gain level settings as well ........... Check out AQ Niagara power conditioners .......... They have several of them ......... They are very favorably reviewed by many audio reviewers ..........
Speaking about EQ, are you familiar with Sonarworks? ........Are you already using their EQ? ......... If not, check them out .......... They have several different EQs for several different 'phones ........ I saw Elear in the list but not Clear (yet) ........... Regarding Rogue RH-5, there are many reviews listed on Rogue Audio website including Stereophile review ........... R.A website also has all the specifications of RH-5 ........
On this page you'll see Audioforge pages 1-10, with thumbnails and full photos of the curves I've created for nearly 200 headphones.
http://dalethorn.com/Photos.html
On this page, among other things, you'll see item 3 Headphone EQ with Audioforge Equalizer, which has the intro and precise settings for the images as noted above. Frequency, 'Q', and amplitude. Below that is item 4, Headphone EQ with Audioforge Tutorial. The tutorial is very brief, but helps a person understand what's involved and what's the goal (i.e. natural sound, NOT correction for individual hearing.)
http://dalethorn.com/Hifi2.html
Thanks for your info ......... I will check those websites as you recommended ............ The poor impulse response of any transducer (mics, headphones, loudspeakers) can cause IMD and TDD (time domain distortion) ......... None of these transducers (so far) have perfect impulse response ...... Ehrlund says their triangular diaphragm (in the mics) has the best impulse response, hence the least amount of IMD and TDD ........... The MQA description of "blurring" could be IMD ...... Of course they also talk about TDD ....... My thinking is ...... are these distortions originating from the microphone? They may not be originating from the digital recording ......... The old computer saying "garbage in garbage out" may apply here ........ THD is different, although the origin of THD could be from the microphone ........ People measure THD+N ......... Is the "noise" originating from the microphone? ......... As I said before, I was trying to connect all the dots ........ Poor impulse response could also cause high Q factor .......... This could be the "ringing" Ehrlund describes ..........
This is such a huge topic it will have multiple answers. First, the ehrlund mic reviews make it pretty clear that they are less a factor in distortions than most anything else. If you have time, check out the IsoMike site and maybe follow its logic to see how the left-right isolation reduces phase and other similar problems. Note that this is NOT specific to a particular microphone - it's a technique, and the Chopin Last Waltz recording is a great example.
http://isomike.com/he2004.html
One of the reasons a parametric equalizer is vital is because of strong narrow peaks and dips in the response, and you need a good continuous tone sweep to be sure where such narrow deviations occur. So even if you have a ringing problem, you may be able to suppress a particular peak in the response where the ringing is worst, enough to make your music listenable and enjoyable without losing that frequency entirely.
I can't guess exactly what MQA is doing, but as long as I have a good parametric equalizer I can make partial corrections for nearly anything, at least until a problem is so bad that a loss of detail in the area of that problem or distortion becomes so obvious that it can't be ignored - and then it's time to replace some hardware.
I like to find good gear at a reasonable price, but I respect the fact that the higher priced gear usually sounds better - not because it's always more neutral and detailed, but because it usually fixes a lot of the issues that cheaper gear can't.
You can use external power conditioners for cleaning up electricity ........ AQ makes several of these power conditioners ........ Also PS audio makes them ....... At least dozen other companies make these power conditioners .......... Many of these including AQ and PS audio are very favorably reviewed in Stereophile ....... RH-5 can put out a lot of "juice" ........ It can drive any 'phone ....... I use Hugo2 which is battery powered (internal supply) ........ It can easily drive a lot of high efficiency low impedance 'phones including my Lcd-x and Lcd-MX4 ....... It has problem driving low efficiency high impedance 'phones like my Lcd-4 ...... Hugo2 sounds great with my high efficiency low impedance 'phones ....... Hugo2 is very favorably reviewed by many (and many "golden ear") audio reviewers ........
Visit the website before you make up your mind, please ....... It is free ....... There are several videos, reviews and comments by several artists and recoding engineers, you can find on Google search ....... You can ask dalethorn above for further information ......... He posted some comments above about these microphones as well .........