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It's sample accurate ?

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No long story about what's wrong with current digital thoughts (as compared to analog transparency and high sound fidelity) , just an observation that had me occupied long ago already.


If you use a program to record digital audio, or to play it back, will it be sample accurate?


Some years ago I recall installing Steinberg - something claiming (finally) their recording software to be sample accurate, which didn't imply 0 depth buffering or sub-sample time controller response (which is quite easy to achieve in dedicated hardware, but not really thinkable on a PC) but simply that the sample sequence you put into, for instance, a track, comes out again unchanged, sample for sample. As in the individual L & R say 24 bit digital numbers that were put into a track, come out the exact same way, from beginning to end.


The same principle for programs reading/writing sample files: Sox (I know it only on Linux) for instance can put off all processing and adaptation filters so it reads a sound file, and writes the exact same sample data, for instance in a different format, with no changes to the sound data. So I'd convert a 16 bit cd quality audio file to a 16 bit, same sample rate .flac lossless encoded file, and that would play back the exact same music data. Which could be verified by stripping the original file to a raw data file with only audio samples, decoding the flac file to a data file as well, and byte-for-byte comparing the data files, which of course are exactly the same size as well!


Boring proposition, perhaps. But not so much so, there are audio processing that change data, I researched it a while ago and for instance audacity changed audio data over a simple read file -> write other file operation. I found at the time (some years ago) audio editor "Sweep" (at least on Linux) does a perfect job of cutting and pasting audio data without any change or length change.


Does it matter? Well, some operations such as streaming the data through a more or less neutrally adjusted FFT processing step also when requesting no processing might seem alright to some, but at some level, and those levels I'm interested in, there's a big difference that makes such a tool useless for my purposes. I require big amounts of subtle filtering that in the end isn't subtle, and very sensitive to all kinds of digital cludging about.


For "normal" use, it's not right when your CD mastering tool changes data even if you tell it to do nothing, IMO. Of course you can *want* to impart some sound impression by using an effect tool, that's an entirely different story, I'm talking about when you don't!



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I researched it a while ago and for instance audacity changed audio data over a simple read file -> write other file operation.


Why do you think that is? The audio engine rounding off certain values?


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No long story about what's wrong with current digital thoughts (as compared to analog transparency and high sound fidelity) , just an observation that had me occupied long ago already.T.


Your OP is an interesting (and not unexpected) read. Your quest for improvements is certainly welcome. Things can and will be improved due to efforts like yours.


I do question this first statement. What analog recording medium and/or playback medium is truly transparent?

Tape inherently has tape hiss, it may be small but it exists. There is a noise floor, all the specs for microphones that I have seen include a self-noise specification so recording is inherently noisy.

The frequencies generated by tape hiss can and do interact with frequencies being recorded, resulting in new harmonics created.

Only a few humans (if any) have the luxury of only listening to first generation tape recordings on the original 2" tape running 30 i.p.s..

I've heard that, once. It sounded amazing.


As transfer media, both vinyl and tape have background noise, adding noise to noise (to noise).

Obviously we've no control over choice of playback but it is likely not to improve things.

I've a friend who spent tens of thousands of dollars on hi-fi gear. It makes a tiny piece of dust in the groove of the record really stand out clearly.

It also makes radio broadcast sound like it sounds, very compressed.


Short of playing live music in a great sounding isolation room on acoustic instruments - which is by far the best sound - how do we distribute this "analog transparency and high sound fidelity"?

I am truly curious and grateful for any answer you may provide.




It took a chunk of my life to get here and I am still not sure where "here" is.
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The question of sample accuracy comes up now and then, but nobody has come up with a good reason for it other than either to prove that either a copy is or is not mathematically a perfect copy. In theory, you should be able to make a copy of a file and every bit of that file will be the same as the source, and in practice they will be. If you do a bit-comparison of the two files, it will "pass." If you can measure any difference in the final output, for example if you sum the original with a polarity-inverted copy and there's something left over, either your summing thingy isn't perfect or there's imperfect timing or level translation in the digital-to-analog conversion.


In real life, has a few bits of inaccuracy ever mattered? It's a "nice to know" thing, but not something to worry about today. When A/D and D/A converters were much more crude than they are today, there was some valid concern. But today, good converter chips are so good that you don't need to spend thousands of dollars on parts and matching in order to do accurate and repeatable conversion - that's part of what made a $2,000 Prism or Benchmark better than a $59 SoundBlaster.


Today the primary difference between a high grade converter and a garden variety one is in the design of the analog end and the things that influence it like power supplies, connectors, circuit board layout, and shielding. There's still a difference between a Prism and a SoundBlaster but it's not as audible as it was 30 years ago.




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For my purposes, I need a lot (I mean hundreds, even a 1000+ "plugins") of accurate processing steps, which work sample accurate on "jack" the Linux audio streaming standard (also available on Windows).


It would be kind of weird to throw away all that accuracy simply to do a simple processing step with an audio editing program...


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