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Thread: Not another debate thread - Just a question

  1. #51
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    Quote Originally Posted by famous beagle
    It doesn't really matter to me either way, because it's easy enough to dither. But to get to Monty's point, it seemed to me that he was saying that, at 16 bits, not dithering is not going to "ruin" a master because the difference would be so hard to hear.
    It's easy enough to dither, and it would be easier if we didn't have to think about it. For a while now, Samplitude has had a thing called smart dither or something that you can enable so the workstation just dithers automatically when necessary. I don't think all DAWs handle it the same way.

    As for the difference being hard to hear, it certainly can be for something like an ABX comparison. Sometimes you can hear things like reverb tails where dithering preserves the length of decay and sense of space better. Without listening for a specific artifact it can be difficult to hear, but once we find something like that it can get much easier. How a signal with truncation distortion will react to further processing is a concern as well. If you don't notice it for 2 or 3 generations, it's too late to do anything about it. We have the option of just making sure the signal doesn't have that crud in it.

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  3. #52
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    Quote Originally Posted by famous beagle View Post
    BUT, there's also this concept called wave particle duality. And it says that everything (every particle)---whether you're talking about light or matter---acts as both a particle and a wave, depending on when/how you look at it. (I'm paraphrasing horribly here.) And so that muddies the waters a bit more. I'm more confused now than ever!
    It's very important to remember that quantum mechanics says literally nothing about what actually is. It is merely a convenient model that helps to predict what might be. It is actually more likely that there is neither particle nor wave, but using those ideas can help us to predict - or more precisely figure the probablity of - certain phenomena coming to actuality. That is, it's all just math and pictures to help us conceptualize and comprehend things that are essentiallly inconceivable and uncomprehensible. I know that doesn't help your headache, but that's where we're at.

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  5. #53
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    Quote Originally Posted by ashcat_lt View Post
    It's very important to remember that quantum mechanics says literally nothing about what actually is. It is merely a convenient model that helps to predict what might be. It is actually more likely that there is neither particle nor wave, but using those ideas can help us to predict - or more precisely figure the probablity of - certain phenomena coming to actuality. That is, it's all just math and pictures to help us conceptualize and comprehend things that are essentiallly inconceivable and uncomprehensible. I know that doesn't help your headache, but that's where we're at.
    Yes ... it really truly is mind-boggling. The double-slit experiment and the Mach-Zehnder interferometer simply blow my mind.
    famous beagle

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    You have a point: The fewer magnetic particles passing the record point per second, the lower the maximum audio frequency you can record. Consider the situation with those open-reel tapedecks of the 50s, 60s, and 70s; and note that they commonly had three speeds at least on consumer decks (professional decks often had 15IPS - really "hauling the tape").

    There is one reason for the availability of the higher speeds - you can record signals of higher frequency if you can move more oxide past your record point per second.

    Regarding the relationship between the analog audio signal and the high-frequency bias signal, one service technician whom I knew in Oklahoma told me that in a way, the analog signal modulates the bias signal as the carrier. And as in AM radio, if you overmodulate the carrier, you get distortion - not as bad as that in digital equipment when the converter is "overmodulated" - but still a kind of distortion you cannot undo in later processing of the recording.

    As you indicated, there are also varying qualities of recording tape. I've seen some cheap tape which has dropouts and just poor quality recording as a result. If you want to make a high-quality recording, it is well to get hold of a reel of quality commensurate with the need for quality of your recording at hand.

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  8. #55
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    Quote Originally Posted by hautbois16 View Post
    You have a point: The fewer magnetic particles passing the record point per second, the lower the maximum audio frequency you can record. Consider the situation with those open-reel tapedecks of the 50s, 60s, and 70s; and note that they commonly had three speeds at least on consumer decks (professional decks often had 15IPS - really "hauling the tape").

    There is one reason for the availability of the higher speeds - you can record signals of higher frequency if you can move more oxide past your record point per second.

    Regarding the relationship between the analog audio signal and the high-frequency bias signal, one service technician whom I knew in Oklahoma told me that in a way, the analog signal modulates the bias signal as the carrier. And as in AM radio, if you overmodulate the carrier, you get distortion - not as bad as that in digital equipment when the converter is "overmodulated" - but still a kind of distortion you cannot undo in later processing of the recording.

    As you indicated, there are also varying qualities of recording tape. I've seen some cheap tape which has dropouts and just poor quality recording as a result. If you want to make a high-quality recording, it is well to get hold of a reel of quality commensurate with the need for quality of your recording at hand.
    Err? Not quite. There are vastly more magnetic domains passing a tape head even at 1.7/8ips than are needed for HF response (I seem to recall a Nakamichi that got to 25kHz?). The problem is output. The output of a system that reads magnetic flux is proportional to the Rate of Change of that flux so the output of a tape head rises at 6dB/octave. Obviously, the higher the tape speed, the greater the output at any given speed. This we get the familiar playback time constants which would be a perfect LP, 6dB oct' but there are inevitable losses in the heads and tape formulations. We can compensate for those to some extent but as tape speed falls we get progressively more HF 'Squash'. So, we have to test cassette response at neg 20 but we can do it at -10 (ref whatever flux standard you are using) for 15ips. The wider tracks help as well but really mostly improve noise performance (signal doubles for twice track width, 6dB but uncorrelated noise only goes up by 3dB)

    Then, there is really little problem WRITING to tape* but the replay head gap puts a limit on HF. So, make finer gaps? They did but then overall output falls. Cassette really is something of a marvel!

    Yes, HF bias must be at least twice HF limit. Sound a bit familiar?

    If Mr Beats is looking in please be gently with me, I learned this stuff nearly half a century ago and have not checked any references!

    *The arrival of the transistor posed a problem in getting enough voltage swing to ensure constant current drive but as circuits developed, amplifiers with current feedback, they were overcome..To an extent, as tape formulations improved and could be hit harder, amplifiers had to play 'catchup' Some replay amps also began to get short on headroom.

    Dave.

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    hautbois16, please see my PM. Couple of things come to mind?
    You say you "like the KNOBS" fine, so I thought you might find a MIDI surface controller handy? That converts much of the DAW visual parameters into knobs and slider movements. I hope the interface you chose has MIDI but if not USB will probably be fine.

    You also mentioned making a "sacrificial track" ? You don't have to bugger a perfectly good one! Just save it and send it to a USB stick or external drive. Then you always have the track saved virgo intacta.

    Another issue was with fekking Windows changing sound settings? I curse this often with Skype as I use a headset into a USB port but at odd times the computer reverts to "High definition Input" instead of "USB CODEC" . I f you are using a desktop you might consider an internal PCI sound card? The M-Audio 2496 can still be found and can be set as default sounds then you disable the internal MOBO sound and just run higher quality through the M-A. Bloody Windows ten then has nowhere else to go!

    Dave.

  10. #57
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    Great thread this! I would just like to add that there is a school of thought that time itself is quantised. It could be that you can't get any shorter a time interval than maybe the time it takes light to traverse a sub atomic particle. That could mean that time jumps from one interval to the next like a series of movie frames to give the illusion of continuity. Sort of puts the cat among the pigeons in terms of sound waves......

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  12. #58
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    Quote Originally Posted by Findlay View Post
    Great thread this! I would just like to add that there is a school of thought that time itself is quantised. It could be that you can't get any shorter a time interval than maybe the time it takes light to traverse a sub atomic particle. That could mean that time jumps from one interval to the next like a series of movie frames to give the illusion of continuity. Sort of puts the cat among the pigeons in terms of sound waves......
    Given that Einstein proved that time and space are inextricably linked, this would make sense to me. It's actually almost hard to imagine that it couldn't be true, assuming that.
    famous beagle

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  14. #59
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    Quote Originally Posted by famous beagle View Post
    Given that Einstein proved that time and space are inextricably linked, this would make sense to me. It's actually almost hard to imagine that it couldn't be true, assuming that.
    Got to be tied up with 'data'? Information Theory (of which I know Jack S) surely means once we "cannot know" THAT is the limit?

    Or! The time taken for a photon to pass a given point?

    Dave.

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