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

 

[SebastianG]

Just for kicks and giggles I rewrote most of "dsd2pcm" in C and C++. Compared to the java tool, it lacks the 2nd stage but it supports multiple channels and user-selectable bit ordering and output bitdepth (16 or 24). Without the 2nd stage you basically get "DXD" (PCM at 352.kHz). I figured, once you have a PCM stream you could just as well use any sampling rate converter to do further processing. Though, keep in mind that the resulting 24/352 PCM signal is a big waste of hard disc space since there will be a lot of ultrasonic noise in there (blame DSD, not me!). I recommend further downsampling and/or an additional lowpass filter (kicking in at 50 kHz). If you like such high sampling rates and want to do lossless compression afterwards, you really should do some filtering.

You might want to download some DSD files from here to test a little. I successfully tested a DSD64 stereo track like this:

CODE

sg@home:~/ccpp/dsd2pcm$ cat 2L38_01_DSD.dff | ./dsd2pcm 2 m 24 | sox -t raw -r 352800 -s -3 -c 2 - -r 88200 -b 24 test.wav

(Of course, this only works for uncompressed DSD files)

This conversion was done in approx 30 X real-time on my computer (2.4 GHz Core2 Duo). So, I'm a little surprized to see some people are having problems playing back DSD in real time. BTW: I didn't find the foobar2k DSD plugin. Does anybody have a link?

Cheers!
SG

latest source code: https://code.google.com/p/dsd2pcm/

This post has been edited by SebastianG: Dec 4 2011, 17:43

[vincefalks]

Just for kicks and giggles I rewrote most of "dsd2pcm" in C and C++. Compared to the java tool, it lacks the 2nd stage but it supports multiple channels and user-selectable bit ordering and output bitdepth (16 or 24). Without the 2nd stage you basically get "DXD" (PCM at 352.kHz). I figured, once you have a PCM stream you could just as well use any sampling rate converter to do further processing.

You might want to download some DSD files from here to test a little. I successfully tested a DSD64 stereo track like this:

CODE

sg@home:~/ccpp/dsd2pcm$ cat 2L38_01_DSD.dff | ./dsd2pcm 2 m 24 | sox -t raw -r 352800 -s -3 -c 2 - -r 88200 -b 24 test.wav

(Of course, this only works for uncompressed DSD files)

That's awesome, SG! I'm going to try a multichannel DFF...super . Oh no hang on - bah I'm a n00b - how do you compile it in windows, or can you provide a windows binary?

Honeslty not much info is known on this, but people keep claiming that 352.8kHz PCM IS DXD, when I really think it's not. I myself don't have tonnes of info on it, but for TRUE DXD I think you need to change something else in the encoding of it too (different roll-off filter to PCM). So it's just ultra hi-res PCM in this case.

This conversion was done in approx 30 X real-time on my computer (2.4 GHz Core2 Duo). So, I'm a little surprized to see some people are having problems playing back DSD in real time. BTW: I didn't find the foobar2k DSD plugin. Does anybody have a link?

Cheers!
SG

It's here.

It's DST-encoded files that we're having troubles with, not uncompressed DSD. The reference decoder is quite "unoptimal" by the looks of it, and could be improved a lot. Right now it's only good for storage, not on the fly decoding.

[SebastianG]

That's awesome, SG! I'm going to try a multichannel DFF...super . Oh no hang on - bah I'm a n00b - how do you compile it in windows, or can you provide a windows binary?

Sorry, I can only offer a "32bit x86 Linux binary" at the moment.

It's here.

Sweet.

Cheers,
SG

Attached File(s)

 dsd2pcm_32bit_x86_linux.zip ( 8.4K ) Number of downloads: 156
 dsd2pcm_32bit_x86_linux_static.zip ( 412.24K ) Number of downloads: 155

 

[vincefalks]

Sorry, I can only offer a "32bit x86 Linux binary" at the moment.

Ok. well do you have a paypal account, maybe I can speed it up

Honeslty not much info is known on this, but people keep claiming that 352.8kHz PCM IS DXD, when I really think it's not. I myself don't have tonnes of info on it, but for TRUE DXD I think you need to change something else in the encoding of it too (different roll-off filter to PCM). So it's just ultra hi-res PCM in this case.

How can a n00b with only 18 posts already cause so much pain? You yourself might not "have tonnes of info on it", but apparently tonnes of bull***** to spread anyway. All you have posted here indicates an urge to find technical justification for marketing induced belief. You don't understand what DXD is, but you know it must be better. You don't understand what DSD is, but hey it must have something to it, even if it is analogish distortion, as reported by your choice of experts.

DXD fixes a bad design choice of DSD (by todays standards) and the fix is plain PCM. But hey, it can't be. DXD is supposed to be better than DSD, DSD is supposed to be better than PCM, ergo DXD can't be just PCM, else it would sound "digitalish", wouldn't it? That is your apparent mindset. When you say you "really think it's not" it would be more honest to say you "really believe it's not" as in "I believe in the maidenhood of mary". I think there are other forums where you would get much more positive resonance for that kind of interest.

I'm sorry, but you're only embarassing yourself. Let me rebut you and completely blow away your silly assumptions with *fact*, *reason* and *science*.

1. I have another alias on hydrogenaudio, which I used up until recently, and did so for about 2, maybe 3 years IIRC. It was nothing like a 300+ post profile and I've never been a "regular" per se, but by me merely being "new" you're assuming that I don't know anything about HA, aren't you? Completely wrong. I have been frequenting and posting at these boards for years - and knopw exactly what this community is about. I switched to this new username for a certain reason which I have a right to keep to myself, and no it has nothing to do with anything I did in my previous profile and "starting afresh" or something stupid like that - as I said, I was never an active user anyway. It's a certain privacy issue (I have changed identities for certain privacy reasons) which I have a right to do. Oh and if you now took this info I've just posted and falsely accuse and assume (once again) that I must have been some idiot and got caught in some embarassing or controversial affair on HA and started afresh to escape that previous identity (it seems you are quick to falsely blame people of that sort of thing), that would just be completely lame. The mods would know, as per the IP addresses under my posts on both profiles, who I used to be, although I recently moved houses and IPs too....

2. Have I indicated ANYWHERE that I believe that certain mathematical characteristics in digital audio (like different roll-off filters), correlates to better/different perceived audio quality? NO! Do I have to justfy why I want a certain setting/feature in digital audio? NO! I might just be someone who wants to edit audio files in the highest possible anal quality (and not for subjective "audio quality" reasons)! It's not your business to question why I am persuing certain digital formats or conversion/encoding settings. I have not claimed ANYWHERE that I am pursuing certain formats for their "audio quality".

You are making an ASSUMPTION based on stereotypes and linking it with a new profile here at HA (I'm not new) and my self-admittance of not being an expert on digital signal processing, and my raising questions which might be n00by and even unscientific in nature (but NOT invalid as I'm not making any claims - "wondering" is a part of science and the unknown), and creating a picture that is completely innacurate!

Can you stop n00b bashing please? Just because I'm not an expert on digitial signal processing....I unashamedly participate in these forums no matter what my level of knowledge is, and someone like you and your attitude, I'm sorry, but that won't stop me from making my contributions, whatever you interpret them to be.

Now back to THIS case about DXD: I am interpreting what seems to be not very widely-known SCIENTIFIC information on a very new digital audio format. All sources of information say that DXD is PCM-like, NOT PCM!!!!

The white paper, and other sources on the internet, say that because it has an ultra-high sampling rate, "it can afford to have a more relaxed anti-alising filter" (and thus, characteristically, has a closer pulse response and frequency response to DSD). This is scientific information, NOTHING to do with pereived sound quality!

Now, I want to get to the bottom of this, because I'm not sure if "it can afford to have a more relaxed anti-alising filter" means it DOES have a more relaxed anti-alising filter than PCM. I READ somewhere that yes, it DOES, but can't find that info since DXD is so obscure and new.

rpp3po, I am neutral and objective in ALL my posts here and you're making a whole lot of claims otherwise!!!

All you have posted here indicates an urge to find technical justification for marketing induced belief

Let me put it to you mate: I have an urge for TECHNICAL PERFECTION (in this case), INDEPENDENT OF WHAT THAT MIGHT MEAN IN TERMS OF SOUND QUALITY! And I don't CARE if you think that's overkill or unnecessary or crazy or a waste of bandwidth or effort or space! I really don't...