DSD-2-PCM -- proof of concept, test sample and source code here Options

[SebastianG]

The ZIP contains 4 files:
- dsd2pcm.jar (the converter written in Java, outputs raw 24/88 PCM, intel byte order)
- info.txt (contains some infos)
- test2822k.dsd (14 seconds, mono DSD, 5 megs)
- test44k.mp3 (conversion result)

have fun,
Sebi

Edit-2011: For the latest C/C++ source code see https://code.google.com/p/dsd2pcm/ It is released under the new BSD license. The attatchment to this post is still the first Java release.



This post has been edited by SebastianG: Dec 4 2011, 17:45

Attached File(s)

 dsd2pcm.zip ( 3.56MB ) Number of downloads: 2498

 

[Axon]

Yes, let's all thank LukeS for being a boy scout and reciting Digital Signal Processing 101 to the rest of us. Particularly that bit about how designing reconstruction filters with a transition band of 0.08Fs is impossible, which is why you can't encode 0-22k perfectly in Fs=48khz. And how FIR filters cause nonlinear distortion.

Sorry for being a bastard, since it doesn't seem like you're wrong per se, at least on most of what you're saying. Nobody's disputing that the transition band ought to be at least a couple khz wide to reduce ringing, or that we'd all like to increase sample rates in the absence of any other cost or effect.

You've read Lipshitz/Vanderkooy, right?

This post has been edited by Axon: Nov 25 2009, 05:18

[LukeS]

Yes, let's all thank LukeS for being a boy scout and reciting Digital Signal Processing 101 to the rest of us. Particularly that bit about how designing reconstruction filters with a transition band of 0.08Fs is impossible, which is why you can't encode 0-22k perfectly in Fs=48khz. And how FIR filters cause nonlinear distortion.

Sorry for being a bastard, since it doesn't seem like you're wrong per se, at least on most of what you're saying. Nobody's disputing that the transition band ought to be at least a couple khz wide to reduce ringing, or that we'd all like to increase sample rates in the absence of any other cost or effect.

You've read Lipshitz/Vanderkooy, right?

No need to be sarcastic, if you feel like explaining it better please do so. I may have messed up the terminology because DSP is not my thing. I am not much of an analog or DSP person, I do digital design and prefer to stick with board design and digital logic design. The bottom line was I knew that 48kHz does not perfectly preserve the 0-22kHz band and the guy wanted an explanation so I tried.

Thanks for the link to the paper, I am really new to SACD and i'll give it a read.

I nowhere claim SACD is perfect either and really just want to rip it for the multichannel audio that many artists do not offer in any other format.

This post has been edited by LukeS: Nov 25 2009, 05:26

[Martel]

The bottom line was I knew that 48kHz does not perfectly preserve the 0-22kHz band and the guy wanted an explanation so I tried.

The question is - how many people actually care? If you have a 48kHz fs content which is has a low-pass applied rolling-off from 21kHz (say -3dB @ 21kHz) how many people would be able to actually hear this? I remember when I was 18 I could hear up to 20kHz (18,5 kHz @ 29) but since most people usually don't have enough money at that age to buy an equipment able to reproduce such content flawlessly, I guess it does not matter much if at all (the consequences being largely academical).
The point is, if you want to hear the "true music", you go to a live performance where the stuff in question does not usually matter.