Can the Quantum be switched in or out?
The 'DSP/OUT' switch enables/disables processing. The 'quantum' word applies to their 5th generation dac name which is included in a few of their products. They just mean the new dac is a quantum improvement over their last dac design. I guess they could have called it 'leap' or 'step' or something, but quantum presumably has a nicer sound to it.
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It would probably sound even nicer if they'd spent a bit more time and money on the aesthetics 😉quantum presumably has a nicer sound to it.
It would probably sound even nicer if they'd spent a bit more time and money on the aesthetics 😉
They are a very reputable company known for excellent sound quality. Many mastering rooms and high end recording studios feature various Crane Song boxes. This unit pictured is designed for rack mounting, not something most people use at home. The color-scheme and look has been a Crane Song trademark for decades, even if it looks like it. Most people keep their Crane Song boxes that long anyway.
I’m finding it hard to comprehend that the differences between those two tracks are caused from adding distortion to what I would have considered a already distorted source?......but I am after all here to learn.
Bob,
Depends on the nature of the original distortion. If it sounded, bland, muffled and or lacking dynamics and detail, a little distortion might help. 2nd order distortion makes it sound fatter, thicker, sometimes richer (maybe like playing a note on a 12-string guitar vs a normal guitar, the octave string is at the 2nd harmonic of the lower string at least for the string pairs on the bass side of the instrument). 3rd harmonic can add a sense of clarity and or detail to something that is muffled sounding and or lacking detail.
That kind of thing can sound good on a single instrument or vocal, but when vocal or instrumental harmonies are played, then the associated Intermodulation Distortion can be heard.
Depends on the nature of the original distortion. If it sounded, bland, muffled and or lacking dynamics and detail, a little distortion might help. 2nd order distortion makes it sound fatter, thicker, sometimes richer (maybe like playing a note on a 12-string guitar vs a normal guitar, the octave string is at the 2nd harmonic of the lower string at least for the string pairs on the bass side of the instrument). 3rd harmonic can add a sense of clarity and or detail to something that is muffled sounding and or lacking detail.
That kind of thing can sound good on a single instrument or vocal, but when vocal or instrumental harmonies are played, then the associated Intermodulation Distortion can be heard.
I can tell you exactly what to do with the bits and bytes (programming-wise) to preserve bit-accurate reproduction of the 16-bit file at a reduced volume in a 24-bit word, but I don't know of an audio editing program that you could do this exact thing with.Yes, I want to start with any file and convert it to 24bit wav file so that I can run loopback recordings of a variety of loopback cables. Many recordings are maximised to 0dB or just under and many of these 'go over' on playback so I find 2dB and sometimes 3dB attenuation is required to avoid record input clipping with precisely fixed unity input gain structure.
So all minds here please, what is committee approved method/software of converting any audio file to 24bit with attenuation of some amount, fixed 3dB would be ok ?.
Once this is established I can move on and test and compare a range of cables with real music waveforms and capture valid differences for analysis.
Drop each 16 bit sample into a 24 bit word. Do a signed shift right one bit of this 24 bit word, effectively dividing by 2.
This will reduce the level by 3.0102999566398119521373889472449 dB (this is 10 times log 2, which exactly converts a 2 to 1 voltage ratio change to dB, calculated with the MS Windows calculator), or basically, close enough to 3dB. You could as easily shift right two bits, giving a 6.02... dB reduction.
The good news is it's easy enough in Audacity or other audio editor to load in a 16 bit file, convert it to a 24 bit file (I think in Audacity I would copy the 16 bit file, create a new 24 bit file, and paste), and then reduce the volume by 3dB. This will NOT shift the bits as would the above scenario, but it doesn't matter. Instead of shifting one bit, it creates a 24 bit word that's 3dB below the original signal, and is a representation of the original that is accurate to 24 bits. When doing this it really doesn't matter what the exact volume reduction is. I might go with 4dB or 6dB to be sure. Any level in the single-digit range will work (at extremes such as -40db the low-level detail gets lost in the reproduction system's noise).
What an audio editor does is a 16bit (the original audio samples) by 24bit (a number representing the volume change amount) multiplication, which results in a 40 bit word that gets truncated (though some audio editors may dither at this point) to 24 bits to store in the 24bit file. If you don't think that's good enough, then you need a program that does the signed shift right to perfectly preserve the original 16 bits as I described above.
Bob,
Depends on the nature of the original distortion. If it sounded, bland, muffled and or lacking dynamics and detail, a little distortion might help. 2nd order distortion makes it sound fatter, thicker, sometimes richer (maybe like playing a note on a 12-string guitar vs a normal guitar, the octave string is at the 2nd harmonic of the lower string at least for the string pairs on the bass side of the instrument). 3rd harmonic can add a sense of clarity and or detail to something that is muffled sounding and or lacking detail.
That kind of thing can sound good on a single instrument or vocal, but when vocal or instrumental harmonies are played, then the associated Intermodulation Distortion can be heard.
Ahh.....much like my aphex aural exciter does in the analog realm?
You are a dark horse Bob, you knew all the time 🙂
Sometimes it’s a conundrum knowing so much but having no clue! 😀
Edit.....well if that’s the case I’d like to see a decent 16/44.1 recording directly run through dans process to hear if it can improve it.
My aural exciter only works to ‘fix’ problems in a live recording during mastering......it definately does not improve a good already mastered recording. (I’ve tried!)
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Devices such as the one Mark described (one of the originals from the 1970s was the Aphex Aural Exciter) have been used to "sweeten up" individual instrument sounds for decades, just as he described. I recall some controversy and consternation in the hi-fi mags back then that "they're actually intentionally ADDING DISTORTION to recordings in the studio!" It was like the auto-tune of its time.I’m finding it hard to comprehend that the differences between those two tracks are caused from adding distortion to what I would have considered a already distorted source?......but I am after all here to learn.
Such a tool also has its use in audio restoration, where a recording might only have a frequency response to, say, 3 kHz. Such a recording sounds dull (like the high treble control is turned down) even to most old folks like me - my hearing drops off around 10 to 12kHz. You can run the original through a high-pass filter around 2kHz, run that through something to create distortion (which generates sounds well above 3kHz), then mix in the original signal. This adds/restores higher frequencies to vocal "sh" sounds and such, making (what many or most people think) a better sounding recording than the original.
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So is at high audio frequencies. That is unless you know of evidence on audible difference.
Think of Feynman, oneself is the easiest to fool ..... (As said before, it helps to follow the context instead of jumping on trigger words)
Spreading FUD again? 🙄
Either that or just an attempt to offer scott wurcer another interpretation of RNMarsh's post. 😉
IOW, you don't know of any evidence that skin effect's impact is audible at high audio frequencies. Got it.Think of Feynman, oneself is the easiest to fool ..... (As said before, it helps to follow the context instead of jumping on trigger words)
Deflection and redirection noted.Either that or just an attempt to offer scott wurcer another interpretation of RNMarsh's post. 😉
Either that or just an attempt to offer scott wurcer another interpretation of RNMarsh's post. 😉
The exact context of JC's claim was that the BQP's removed TT rumble and other low frequency noise without affecting the low frequency content of the music. This claim directly violates the second law and is extraordinary, the fact that supposedly technically minded members of this forum are trying to figure out how this can be is also extraordinary.
I'll say again, it insults the intelligence of many here. Talking about propagation velocity of 2" of wire is silly even more silly because the propagation velocity inside a piece of wire is somewhat meaningless.
I would not use Audacity in windows for "critical" file making. There are enough instances of Audacity NOT doing what you ask it to do to give concern. If you do use Audacity you need to verify that the format of the output file is what you expect it to be and that the data in it conforms to what it should be (instead of being padding or some such activity) I believe its mostly driver problems and windows (particularly 10).
Reaper is probably a better bet, and its free to use as well, just more complicated. Use with ASIO drivers.
have fun
Alan
Reaper is probably a better bet, and its free to use as well, just more complicated. Use with ASIO drivers.
have fun
Alan
. . . and of course therein lies how the Bybees work.The exact context of JC's claim was that the BQP's removed TT rumble and other low frequency noise without affecting the low frequency content of the music. This claim directly violates the second law and is extraordinary, the fact that supposedly technically minded members of this forum are trying to figure out how this can be is also extraordinary.
I'll say again, it insults the intelligence of many here. Talking about propagation velocity of 2" of wire is silly even more silly because the propagation velocity inside a piece of wire is somewhat meaningless.
The resistor heats up as current flows through it. This is turn activates the catalytic coating on the foil. The coating itself (more about this later) is mined from the dunes in the south polar region of Mars through a confidential process known only to Bybee, the CIA and NASA who've been conducting secret robotic operations on the Red planet for the last 30 years (NASA explorer missions were simply undertaken as a diversion it should be noted). This secret ingredient i.e. element, is called Bybeelonium - it has NOT been added yet to the periodic table as it is classified (up until me making it public on this forum, at great personal risk I might add). It should be noted that this material was originally going to be called Martianolonium, but the CIA management team, with the full backing of Lyndon Johnson and all subsequent Presidents, insisted it be confidentially named in honour of its discoverer's theoretical predictions about its fundamental properties, which have I understand been proven correct, but the details as you will fully appreciate remain classified at this point.
The unique property of Bybeelonium (also called 'Baloneyium' for short as is, for example done with 'aluminium' and 'aluminum') ) is that it is able to clearly distinguish the difference between non-asynchronous, non-random information (e.g. so called 'high-end music') and random noise. Noise interacts with Bybeelonium's upper valence bands only, absorbing random energy fluctuations, which in turn further heat the core 25mOhm 'catalytic activator' resistor and further activate the aforementioned catalytic coating in a so called virtuous noise reduction cycle ('VNRC'). The result is noise reduction that cannot be measured by any known instrumentation (and of course, thus not detectable by the enemy), but will deliver airy high's to any modern audio system whilst simultaneously blocking enemy noise bot attacks. It should be noted that the prime reason for the product's development in the first place was to secure America's - and the West's in general - digital systems infrastructure from malevolent digital forces ('MDF' in CIA parlance). That it came to be commercially available to the music and high-end fraternity simply speaks volumes to the USA's commitment to the arts and art appreciation.
Unfortunately, Bybeelonium has been shown not to trap 1/f noise, but ongoing CIA and NASA funded research points to a possible breakthrough in this area of investigation. Bybee was awarded the Nobel prize for Physics for his discovery in 1969, but the offer and announcement were intercepted by the CIA and the prize diverted to Murray Gell-Mann instead in order to maintain secrecy. Naturally, under the threat of being expelled from NATO, the Norwegian authorities capitulated, and the rest is history. In the spirit of truthfulness, transparency and for the good of all mankind, hopefully with this public announcement (again, made at great risk by this writer), the authorities will see fit to retract Gel-Mann's Nobel award and transfer to it rightful recipient in the coming months.
Clearly, in releasing this information, this writer has taken great risk, and no doubt incured the ire of the US goverment and in all likelhood will face prosecutorial action from the CIA and other as yet unnamed Western government agencies. However, it is important that in these maters, the truth prevail.
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The exact context of JC's claim was that the BQP's removed TT rumble and other low frequency noise without affecting the low frequency content of the music. This claim directly violates the second law and is extraordinary, the fact that supposedly technically minded members of this forum are trying to figure out how this can be is also extraordinary.
Seems that I've missed that, but nevertheless RNMarsh's post (yesterday starting this subsubtopic) was about filtering (maybe) at higher frequencies (end of audioband and above).......
Bonsai, very entertaining but I suspect you have a similar intoxication level to me.
Please return to your rational arguments. 😉
Please return to your rational arguments. 😉
I would not use Audacity in windows for "critical" file making. There are enough instances of Audacity NOT doing what you ask it to do to give concern. If you do use Audacity you need to verify that the format of the output file is what you expect it to be
Exactly.
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