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Badly drawn waveforms vs. the audio that’s actually output—filters etc, Split from: Jplay - just another scam? Topic ID: 92856
Wombat
post Feb 14 2012, 16:37
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QUOTE (JimH @ Feb 9 2012, 16:44) *
Cross posting a jplay thread on the computeraudiophile forum. I finally had to say something last night.

http://www.computeraudiophile.com/content/...#comment-127258

I used the M word.

I just did read over the thread at computeraudiophile.
Seems like all comes down to the better RAM handling changing the sound to the positive. So the claim is that inside the same plattform using jplay changes the way even a asynchronous attached device gets its buffer filled.
This at least should be to measure but isn´t for some reason.
My understaning of this implies that every single processor/RAM/platform implementation must sound different which is a problem.

This of cause has to lead to only one possible conclusion: PC based sound playback doesn´t work and should be considered a dead end, there are to many variables at play smile.gif

When i imagine myself to be a software developer like you JimH, that did spend endless hours of his life to develop a complete suite that grew over many years i´d really had to wonder. Isn´t it much easier to code some esotheric code that prevents bit-rot* and demanding twice the money for it?

When you have some time and read over the forum there are several threads that have strange reasoning. This one for example:
http://www.computeraudiophile.com/content/Why-2496-not-24192
They even use scientific pictures wink.gif
It must be said there are of course several people knowing their stuff.

* bit-rot is a term i learned lately and describes the audiophile problem that happens on computer playback all the time
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andy o
post Feb 14 2012, 17:20
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Is it possible to explain non-mathematically (or just a little) why this linked from Wombat's link is wrong? I know it is, just don't understand it very well.

This is one of the pictures accompanying the article:



QUOTE
I tested some 10 kHz and 20 kHz sine waves that were recorded at several word lengths (16 bit or 24 bit) and sampling frequencies (44.1 kHz, 96 kHz, 192 kHz), analyzing them in a software sequencer.

Isn't that just Audacity? I wonder if they "analyzed them in a software sequencer" to sound more professional.

This post has been edited by andy o: Feb 14 2012, 17:23
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Wombat
post Feb 14 2012, 17:47
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Hehe, shortly after posting here there was a nice picture posted that tells the whole story smile.gif
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icstm
post Feb 16 2012, 11:56
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QUOTE (Wombat @ Feb 14 2012, 16:47) *
Hehe, shortly after posting here there was a nice picture posted that tells the whole story smile.gif
where is the nice picture? That is really important to share!
The whole point of FFTs is that using if I am drawing what the FFT has stored in the time domain (ie if I am drawing sine waves) then I only need a couple of points to draw a PERFECT reproduction!
That is one of the biggest issues that people need to first get their head around when understanding digital audio.

when you see an irregular wave, what you need to ask your self is "what regular wave would I have to add together (ie what frequencies would have to be simultaneously present) to create the irregular wave I see”

so an irregular sine wave is in effect a regular one augmented with higher or lower frequency wave to varying amplitude.

Each of the additional waves only needs a couple of points to represent their frequency. As per usual it appears that CA article really doesn’t get what the Nyquist theorem is saying…

However it is right when it says that it is possible for a perfect 8kHz signal sampled at 96kHz to sound better than sampled at 192kHz if the DAC has trouble with 192kHz (ie is a cheaper DAC). The whole point is that a perfect DAC will produce a perfect 8kHz signal when sampled at 16kHz. But the quality of the output has little to do with the conversion and more to do with the analogue output. One of my earliest posts on HA was asking about DACs in AVRs as I am keen to understand the practicalities on consumer grade stuff.

QUOTE (my earlier post)
What determines a good DAC?
I assume the main issues are post the conversion to analogue and that these issues are the usual ones that impact an amp (pre or power), such as clean power supply, suitable analogue filters, etc.
However are there any differences in the digital part of the amp (such as up/over sampling circuits, or the conversion itself)?
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