[LAD] Re: Direct Stream Digital / Pulse Density Modulation musing/questions

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To: <linux-audio-dev@...>
Date: Tuesday, September 25, 2007 - 4:41 am

On 9/24/07, linux-audio-dev-request@lists.linuxaudio.org

> Date: Mon, 24 Sep 2007 21:59:01 +0200

Why is it wrong? If I drew some dots on a waveform and then connected
the dots, to try to reconstruct the waveform, wouldn't I get a better
result with more dots?


my ears tell me that... that's all; it's just subjective. haha, I see
subjective reports don't get you far around here.


well, what do you mean by better? It seems like 24 bit is already
better in terms of dynamic range at any sample rate, but if you mean
more detailed representation of a waveform (in time), it seems like
you necessarily need to have the highest possible sample rate.

Like, if I were just recording an acoustic guitar and vocals, of
course 24 bit would be the best choice.

But if I'm recording a live band, there is just so much stuff
happening at once... You can't pinpoint an exact time when the
keyboard player presses the key, and you can't pinpoint just when I
pluck that bass string. A 96 khz 24bit system might say that the two
events happened at exactly the same time, when really it was closer to
1/100000 of a second apart. Now think about how many times something
like that could happen in a live recording with many instruments and
vocals and background noise from the crowd, etc. I'd rather have the
detail than the dynamic range in that case...


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Messages in current thread:
[LAD] Re: Direct Stream Digital / Pulse Density Modulation m..., Maitland Vaughan-Turner, (Tue Sep 25, 4:41 am)
Re: [LAD] Re: Direct Stream Digital / Pulse Density Modulati..., Bearcat M. Sandor, (Tue Sep 25, 4:55 pm)
Re: [LAD] Re: Direct Stream Digital / Pulse Density Modulati..., Gordon JC Pearce, (Tue Sep 25, 10:21 pm)