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About tmtomh

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  1. Noting that obdurate means, "stubbornly refusing to change one's position or course of action," the term could quite easily be applied to you, Paul. Surely you must realize that the fact that people found the sound was better when MQA was turned on, is not the same as the claim that MQA makes music sound better, if as you note additional changes in the signal path also are taking place when that MQA "on" switch is engaged. The point - which you surely know despite your obduracy in your mode of discussion here - is that if a switch makes the music sound better, and that switch is claimed to be an MQA-ON switch but really is a switch that turns on MQA and makes other changes to the signal path, that's deceptive and does not count as empirical evidence or measurements on behalf of MQA. I know you know this because you have been quite clear that MQA is not demonstrated in isolation. What you're obdurate about - and I really don't understand why - is your refusal to acknowledge that this "not demonstrated in isolation" aspect is precisely what is making mansr and others object to the test results as examples of measurements of MQA.
  2. When you say you use a VNC app, why not just use Mac OS's built-in Screen Sharing? Also, any screen sharing/vnc on a Mac mini is going to be very slow if there is no monitor connected, because when the mini runs headless it does not use its build in video card/GPU. You can remedy that by purchasing a $10 fake/dummy HDMI plug from Amazon. This fools a headless mini into thinking it has a monitor attached, turning on the GPU and providing the necessary graphics acceleration to make VNC/screen sharing usable.
  3. Excellent piece, and most valuable for its reframing and overall perspective. Thanks!
  4. I've just finished (mostly) initial setup of a new listening space, and I would concur with those who say it's very difficult to use acoustic treatments on bass frequencies below about 150Hz. For the 43 and 90Hz humps I think you have to set your expectations relatively low in terms of taming them. For the 110Hz hump it's more feasible but still difficult. I would concerting with the recommendation for the GIK soffit traps - they're more effective than the tri/corner traps simply because they have a lot more depth across a larger surface area - and depth (plus some air gap between the trap and the wall if possible) is pretty much the only thing that matters - the deeper/thicker the trap, the better the absorption of low frequencies. The good news is that even though it's nearly impossible to achieve full absorption of bass reflections, the thicker solutions will still provide some absorption, and even if you can get a hump to go from, say, +20dB to +17dB, that's a real difference and a noticeable improvement. Finally, one free and easy way to smooth out lower-bass response is counter-intuitive: move the speakers closer to the front wall. The shorter the distance, the higher the "1/4 wave" frequency - in other words, with speakers closer to the front wall, the most problematic frequency is higher, and therefore easier to treat with absorptive panels behind the speakers (this is why some placement guides recommend putting the speakers either as close as possible to the front wall, or else at least 3 feet out).
  5. There are multiple ways to measure a noise floor - the traditional SNR as noted above, and the THD-N/SINAD as I'd noted previously. Moreover, the effective noise floor of CD is -120dB with noise-shaping dither. I'm really not trying to win an argument for the sake of winning an argument. My point simply is that one has to go to some lengths to put together an equipment chain that can top the effective noise floor of a properly produced 16-bit CD/digital file, and therefore it's not accurate that 16-bit's noise floor is generally above that of most equipment, even most decent equipment. I am hopeful that in the future, the specs of the very best Hypex-based amps like the Benchmark, along with the specs of the very best DACs, will become more common, especially in more affordable equipment. But for now, the majority of amps and DACs don't have noise floors that reliably or significantly exceed that of properly dithered 16-bit sources. (And that's putting aside the fact that it's almost impossible to be in a practical listening space where a -96dB signal can be heard when the volume level is set so that peaks are not at ear-damaging levels.)
  6. The Benchmark amp is an excellent piece of kit, but I would not necessarily believe -132dB: https://www.audiosciencereview.com/forum/index.php?threads/review-and-measurements-of-benchmark-ahb2-amp.7628/
  7. Well, the noise floor of 16-bit is -96dB (putting aside that -120dB is achievable in practice with the use of noise-shaping dither). If -96dB were 20% above the noise floor of most equipment, then most equipment would have a noise floor of -115 to -120dB (the former being 96 x 1.2 and the latter being 96 divided by 0.8). Based on measurements of a lot of equipment at sites like audioscience review, very few DACs and virtually no amplifiers are capable of -115dB and pretty much none are capable of -120. And the vast majority of amps fall short of even -96dB, while for DACs -96dB is middle of the pack.
  8. As someone who's strongly anti-MQA, I have zero problem with what you say here. Much of MQA's lossiness is in the area of ultrasonic frequencies, and the rest of the damage it does to the source material is in the 16th (15th?) bit. I don't like any of that in principle, but I have zero problem believing that MQA's effects could be inaudible for either the better or the worse in many, many listening situations. Even if MQA makes 24-bit files into 17-bit files, and even if it mangles the ultrasonics, and even if makes 16-bit signals into 15-bit ones, you're still dealing with a noise floor of 90dB or more, and you're still dealing with DA conversion in the 20Hz-20kHz range that is 99.9+% the same whether it's MQA or PCM. So while I believe some folks can hear differences, I also believe others can't - and I believe that the same people might or might not hear any difference depending on equipment, source material, listening volume, room, and so on.
  9. This aspect of your argument I fully agree with: Overspec'ing makes total sense when recording, mixing, and mastering. In particular, I have no problem believing that when it comes to analogue electronics (including transducers like mics and speakers), linearity in the upper audible range might be correlated with ultrasonic frequency capability - in other words, a mic spec'd to 30kHz might be more linear in the highest audible octave (10-20kHz) than a mic spec'd to 20kHz. I don't think it's going to be true 100% of the time with every mic, but I think the chances are good that it will be true often enough to make it worthwhile to use the higher spec'd mike in a professional situation. Respectfully, I have to partially disagree with the first sentence of your comment here (although I certainly agree with your second sentence). I disagree because I don't think analogue equipment with ultrasonic capability is the same as higher digital sampling rates. Once you have a high enough sample rate to accurately encode 20kHz plus sufficient headroom for digital filtering, I don't see any evidence that higher-res sampling makes any audible difference or any difference in accuracy of capture. I have seen many, many digital filter measurements on well-designed, modern DACs, which show that the filters can effectively suppress aliasing without significant phase effects even with a 44.1kHz sample rate - in other words, the filters are steep enough that they can roll off the response sufficiently between 20kHz and 22.05kHz. But I would agree with you that ideally a 48kHz sample rate is better, because it almost doubles the ultrasonic headroom for sampling, giving from 20kHz o 24kHz to do it. But I've yet to see any compelling argument for end-user music files with sample rates above 48kHz. When recording, mixing, and mastering, sure - do it a 96k or 192km, just like using 32-bit float for the bit depth is advisable during recording, mixing, and mastering. But once the thing is done, I have zero problem with downsampling to 24/48k for the final product and based on the evidence I have seen (and heard) it does not negatively impact the sound.
  10. That's quite eye-opening.
  11. Actually, you make a good point there. I made that remark out of frustration and should not have used the term moronic.
  12. Then why bother responding? And why expect any level of civility or respect in others' responses to your comments? Why participate in a discussion forum if you don't give a damn what others say?
  13. No disagreement with any of this. But at the risk of being pedantic, I do feel compelled to note that the argument between jabbr and I is going on because one of is is specifying the conditions under which we are making our statements, and the other is not. I've stated repeatedly that downconversion does remove data (sample). I've stated repeatedly my personal preference for preservation of full/original resolution master sources. I have noted, repeatedly, that redbook downsampled from higher-resolution originals is indeed lossy but that IMHO - and I have always noted IMHO for this piece - it is unproductive and misleading to speak of redbook as lossy in the same way we speak of perceptual encoding (mp3, AAC and so on ) as lossy. No one has to agree with my position here. But the argument persists not because jabbr is disagreeing with my claims, but rather because he is steadfastly ignoring most of them.
  14. That's a common-sense position. But your supposition that I "couldn't care less" about the loss of these masters is, franky, moronic. I sy moronic because the linked article - and once again, are you even reading the things you link to and respond to? - is about a fire that consumed mostly analogue master tapes, which has nothing to do with what we are discussing. Moreover, wanting high-res digital masters to be preserved is a strong preference I absolutely share with you - but your and my shared preference has absolutely nothing to do with whether or not CDs are lossy. You have claimed they are; it has been demonstrated clearly that they are not; and yet you refuse simply to acknowledge that. Finally, it must be stated that in the case of digital masters that exist in 16-bit, 44.1kHz format - which to my understanding is the vast majority of digital mixdown masters produced up until the late '80s/early '90s - a CD is in fact a bit-perfect digital copy that not only is lossless compared to the original, but also contains the exact same data as the original in the exact same resolution. In fact, one could reasonably speculate that a 30 year-old, well-manufactured CD might today be a more reliable storage medium for that 16/44.1kHz digital data than a 16/44.1k master that exists in the form of a U-Matic tape.
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