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DSD-2-PCM -- proof of concept, test sample and source code here
kode54
post Dec 1 2009, 19:55
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@spoon: I presume you modified the stage 1 filter to reverse the bit order, since DSDIFF is most-significant-bit first.
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spoon
post Dec 1 2009, 20:15
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I thought it was a command line option on dsd2pcm?


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krabapple
post Dec 2 2009, 06:56
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This is all peachy for those recording DSD files or tapping into their SACD player DACs, but could one of you gurus please build an SACD ripper for the rest of us? Thanks.

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spoon
post Dec 2 2009, 09:55
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I might be wrong, but I do not think there are any PC compatible drives which can read the SACD layer - I know Sony did a laptop once which could write them. Hence why people are resorting to tapping into the binary data stream inside a SACD player.


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Martel
post Dec 2 2009, 10:35
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QUOTE
There are three types of SACDs[9]:

* Hybrid: The most popular of the three types, hybrid discs include a Red Book layer compatible with most ordinary Compact Disc players, dubbed the "CD layer," and a 4.7 GB SACD layer, dubbed the "HD layer." [10]
* Single-layer: Physically a DVD-5 DVD, a single-layer SACD includes a 4.7 GB HD layer with no CD layer.
* Dual-layer: Physically a DVD-9 DVD, a dual-layer SACD includes two HD layers totaling 8.5 GB, with no CD layer. It enables nearly twice as much data to be stored, but eliminates CD player compatibility. This type is rarely used.


There's still some hope to modify a DVD drive firmware so it can at least recognize the media and then use some sort of low-level raw bit reading.

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.halverhahn
post Dec 2 2009, 11:38
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Just for your information: Wheatus is offering The Lighting EP as DSD download.


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2Bdecided
post Dec 2 2009, 15:11
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QUOTE (.halverhahn @ Dec 2 2009, 10:38) *
Just for your information: Wheatus is offering The Lighting EP as DSD download.


QUOTE
DSD... This is as good as vinyl, or some say, better. ... They far exceed the quality of DVD and Blu-Ray Audio ... stick it in your PS3 and hit play

...better than 88.2kHz 24-bit FLAC apparently. rolleyes.gif


So, to summarise the world as it's seen...
The people who think they want something better than mp3 is small.
The people who want something without loudness-wars-style clipression is small.
The people who want something better than a good CD is even smaller.

...but for that minority, we can now start a new virtual format war - downloaded hi-res FLAC vs downloaded DSD vs downloaded DXD.

Because confusing the heck out of the listener is such a great way to succeed.


...and since we're delivering so many bits, it must be the best you can get - so for most releases we can just forget about surround sound - and we can certainly forget about doing surround sound properly!


Yet another great way for the audio industry to continue its implosion.

Cheers,
David.

EDIT: the sarcasm is obvious above, isn't it? I mean, I don't need to go through and add a " rolleyes.gif " to the end of every line, do I?

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krabapple
post Dec 2 2009, 17:23
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QUOTE (spoon @ Dec 2 2009, 03:55) *
I might be wrong, but I do not think there are any PC compatible drives which can read the SACD layer - I know Sony did a laptop once which could write them. Hence why people are resorting to tapping into the binary data stream inside a SACD player.



Yeah, there is that. And the whole DRM thing. And the fact that SACD is basically moribund. But here's to hoping. wink.gif
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kode54
post Dec 7 2009, 15:32
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QUOTE (spoon @ Dec 1 2009, 11:15) *
I thought it was a command line option on dsd2pcm?

I didn't see all those posts, so the only version I knew about was the original Java code, which only supported least significant bit first.
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vincefalks
post Dec 7 2009, 17:55
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Ok finally my warning was lifted after a little situation was addressed wink.gif .....


Thanks for your information (2Bdecided and rpp3po). Very much appreciated. That's what I'm here for, I got a lot to learn, and I appreciate you sharing your knowledge, which should never be discouraged here (within reasonableness ofc - if it's something that's been discussed to death it would be annoying). Someone like me should not be scared to ask questions in general (and usually it's ones based on uncommon topics like DXD which have certainly not been discussed hardly at all before) and I don't feel that way anyway.

QUOTE (2Bdecided @ Nov 27 2009, 10:59) *
The problem is, unless you've found better sources than me, these sources are just marketing.

e.g. these are sources that also show SACD pulse response being almost infinitely short, while CD is really long (a sinc pulse). This is all-but-nonsense because to get that pulse response from SACD, you have to remove the output filter. If you remove the output filter, you have full scale digital noise - any pulse in there is invisible*. Run this unfiltered signal through an amp and speakers and they'll blow up.

* - so how did they generate these graphs? I can think of three possibilities:

1. They simulated DSD sample frequency with a multibit signal and no noise shaping.
2. They averaged millions of "real" 1-bit DSD "pulse" signals in the multi-bit or floating point domain to remove the noise so that you can actually see the pulse.
3. They drew it with a pen.

I'm not getting how the output filter (a low-pass filter in the analog domain) lessens/makes moot the analog-like impulse response of the digital signal feeding it?... And I'm not sure that the pulse response needs to apply to the ultra high frequencies anyway? If you filter out the noise, you still have the narrow pulse response for the audible band, right?

Secondly: Not that it's a substitute for my own thoughts, and not that I'm defending DSD/DXD (trust me, I'm not, but by the same token I'm not on the pcm-ONLY side either, I'm 100% agnostic and investigative), but this reminds me a bit of the AES papers back and forth about DSD like this and this). Scientifically peer-reviewed papers (am I right in saying that? Are these papers not necessarily peer-reviewed by objective scientifically-minded audio engineers?), arguing against each other (if not well then I'd like to remind, not that it's hugely relevant to my main point right now, about these two scientific studies which plainly contradict each other...). I'm not sure this is _just_ a matter of marketing (which yes I acknowledge, and when I see some of it I laugh too), and what the DSD proponents say, vs. the truth. E.g. in the AES paper defending DSD, they have quite an accurate-looking graphs of impulse response. Is this AES article misleading marketing too? It doesn't look like it's hand-drawn to me.

Btw, I'm not saying higher impulse response is important anyway, just that it was there in the first place and that it appears to be higher in DXD as well...

- - - - -

Now to the DXD matter:

Thanks for the info and those spectrograms!

What I wonder, is how much noise 24-bit 352.8kHz "normal" PCM would have (whatever "normal" is), compared to these 2L "DXD" PCM files? And also, would there be a difference in impulse/frequency response?

The question is, is DXD just a marketing term, and there’s no strict guidelines on what can be called DXD, or does DXD actually have a "format specification" (so to speak), governing exactly what sort of anti-aliasing filter it must have (to be a true DXD encoded signal)? This is why I questioned the claim that a 24-bit 352.8kHz output file from DSD2PCM is a "DXD file".

Of interest is also whether the architecture of the ADC in question has an effect on how much noise there will be....Not that these "optimal" ADCs in the pyramix, AX24 etc DAWs produce low-noise 352.8kHz files (lol). But I once read that the higher the sampling rate, or at least when you start to go above 96kHz, some noise/distortion starts to occur - but that there have however been "advances" in high sampling rate ADC architecture whereby this noise is somewhat reduced - eliminating some past criticisms of 192kHz ADCs. And that the AX24 (one of the few DXD-capable ADCs) is one of those ADCs. Is this correct?

So in the end, the question is whether (true) DXD can only be generated from specific alti-alias settings which match that in the architecture of DXD ADCs manufactured by Digial Audio Denmark (afaik they're all made by the same company, just re-branded into the other DAWs).

QUOTE (krabapple @ Nov 29 2009, 03:56) *
Any high-SR PCM allows for more relaxed filtering compared to Redbook.

DSD (1bit/2.82MHz) was initially meant as an archiving format, not a production format. Then some genius decided consumers should be offered DSD in a delivery format (SACDs). But you can't do common production moves -- e.g., mixing, edits-- in DSD. You have to go to PCM. Hence kludges like intermediate conversion to DXD (i.e., 24bit or 32bit/352.8kHz PCM) by Pyramix, or Sadie's 'DSD-Wide' (waggishly called 'PCM Narrow') which is 8bit/2.82Mhz PCM, followed by final (re)conversion to DSD.
Note that all of the SRs are integer multiples of Redbook's 44.1 (1fs). DSD (8fs) itself was designed to be easily transcoded to PCM in the first place.

The final amusing aspect to all this is that Scarlet Book spec recommends that SACD players incorporate some low-pass filtering (either 50 kHz or 100kHz) to lessen the amount of ultrahigh frequency content the downstream gear would have to (perhaps badly) deal with.

Btw, DXD is not 'new', it's been around since at least early 2004. DSD-Wide's been around even longer.

DXD in the end, is not meant as an editing format for DSD recordings, but as an original master recording format...

They like to record DXD for SACD production because it has almost the same impulse response and frequency response to DSD (within the guidelines of what is "good enough" to them - they say the benefits of DXD outweigh the slight inferiority to DSD in the pulse/freq response departments, and that it sounds better anyway...:|).

And when you compare DXD to PCM/DSD, it IS quite new. Especially in terms of exposure, it's very "new". That's what I meant there.
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Martel
post Dec 7 2009, 18:26
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A (square analog) pulse has an infinitely wide spectrum composition. If you cut off the higher frequencies, the edges of the pulse will start to curve and oscillate.

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krabapple
post Dec 7 2009, 18:26
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QUOTE (vincefalks @ Dec 7 2009, 11:55) *
QUOTE (krabapple @ Nov 29 2009, 03:56) *

Any high-SR PCM allows for more relaxed filtering compared to Redbook.

DSD (1bit/2.82MHz) was initially meant as an archiving format, not a production format. Then some genius decided consumers should be offered DSD in a delivery format (SACDs). But you can't do common production moves -- e.g., mixing, edits-- in DSD. You have to go to PCM. Hence kludges like intermediate conversion to DXD (i.e., 24bit or 32bit/352.8kHz PCM) by Pyramix, or Sadie's 'DSD-Wide' (waggishly called 'PCM Narrow') which is 8bit/2.82Mhz PCM, followed by final (re)conversion to DSD.
Note that all of the SRs are integer multiples of Redbook's 44.1 (1fs). DSD (8fs) itself was designed to be easily transcoded to PCM in the first place.

The final amusing aspect to all this is that Scarlet Book spec recommends that SACD players incorporate some low-pass filtering (either 50 kHz or 100kHz) to lessen the amount of ultrahigh frequency content the downstream gear would have to (perhaps badly) deal with.

Btw, DXD is not 'new', it's been around since at least early 2004. DSD-Wide's been around even longer.


DXD in the end, is not meant as an editing format for DSD recordings, but as an original master recording format...


DXD in the beginning, was created because most editing in the digital domain was impossible in DSD. Using it directly as a recording format instead of going form DSD to DXD in production is an obvious step since it removes one transformation. But the primary reason for its use instead of DSD is that it allows editing while more closely approximating DSD than, say 196 kHz PCM. It can be easily upsampled to DSD for SACD releases.

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2Bdecided
post Dec 7 2009, 18:33
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QUOTE (vincefalks @ Dec 7 2009, 16:55) *
I'm not getting how the output filter (a low-pass filter in the analog domain) lessens/makes moot the analog-like impulse response of the digital signal feeding it?... And I'm not sure that the pulse response needs to apply to the ultra high frequencies anyway? If you filter out the noise, you still have the narrow pulse response for the audible band, right?
Time and frequency are inextricably linked. The pulse length is inversely proportional to the bandwidth. The pulse that's shown for CD - that's the shortest pulse you can get if you filter out everything beyond the audible band. Or, to put it another way, that is a "narrow pulse response for the audible band".

If you remove high frequencies, the pulse response is longer / slower / fatter.


QUOTE
...Are these papers not necessarily peer-reviewed by objective scientifically-minded audio engineers? ...
No. AES conference papers are not peer reviewed. They don't even ask to see the paper before accepting you (just the abstract).

btw, there was a typo in you URL - the pro DSD paper was...
http://tech.juaneda.com/en/articles/dsd.pdf

The graph there (figure 9) seems real. The text on the previous page explains it carefully, and I'm fairly sure I would get the same if I repeated the simulation.

It compares a reasonably sharp (but not brick wall) symmetric (linear phase) FIR filter with a gentle, possibly not symmetric filter.

The graph clearly shows the noise of DSD that gets through the filter. The noise is about 30dB down, but still easily visible. A 30dB SNR is nothing to be proud of.

The text says "as the noise floor contains only high frequency components which are uncorrelated with the audio, they are not perceptible."

The 48kHz PCM audio contains "ringing at a -30 dB level approximately 1 ms before the click, which is very audible."

So ultrasonic noise is inaudible, while ultrasonic ringing is audible?! What a strange conclusion - one without a shred of psychoacoustic evidence to back it up. No psychoacoustic theory to support it. No listening tests to verify it.


Worse still, with a 192kHz sampled system, you could make the filter ringing far less severe than what is shown - and of course, you'd never have the problem of ultrasonic noise.




More generally, the pro-DSD paper was produced by someone employed by a company selling DSD, while the anti-DSD paper was produced by academics.

The anti-DSD paper is technically sound, from two fantastically well respected researchers in the field of digital audio. (The author of the pro-DSD paper is also otherwise well respected AFAIK).

The authors of the anti-DSD paper would readily admit that all the artefacts they reveal should be inaudible - but since the benefits of DSD should also be inaudible, that's hardly an excuse!


QUOTE
these two scientific studies which plainly contradict each other...)
Already discussed ad infinitum here on HA.

QUOTE
What I wonder, is how much noise 24-bit 352.8kHz "normal" PCM would have (whatever "normal" is), compared to these 2L "DXD" PCM files?
If you restrict your self to real world signals, it will have as little noise as the most noise-free signal you can find. Theoretical signals: -144dB FS. With a practical converter, it'll have the noise of the converter - assuming the quietest real world signal will come from short-circuiting the inputs of the converter with copper wire.

This isn't a silly answer. Even a single resistor "has" more noise than a 24-bit digital audio signal. It really is overkill.

Cheers,
David.

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spoon
post Dec 8 2009, 10:21
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Did anyone manage to decode a .dsf file? (such as one of the linked wheatus ones?).


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BOBCHEWIE
post Jan 3 2010, 12:14
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QUOTE (rpp3po @ Nov 24 2009, 10:23) *
Raw DSD means saving about 3 GB per 70 minute stereo album. Kind of a waste when you can save 48kHz files, that should perfectly* preserve the 0-22 kHz band, and only need about 475 MB (FLAC). That's over 2.5 GB wasted, not for discarded but inaudible recorded information but plain quantization distortion, added by an inferior form of digitalization. Why save 2.5 GB of garbage per album? Has there ever been just one solid positive blind test for DSD?

* talk about differences of -110db or lower

ok i have idea lets make or convert our audio to 11khz mono and knock it into mp3 format..after all, its about quantity over quality isnt it?
the next time i read an article that tells me the audio customers arent interested in quality and the SACD /DVD Audio format is dead ..i will
scream, have a hissy fit and take my ball back...why the heck do we have to put up with this 'anti hires' propaganda all the time..what ?so that el cheapo music company can keep on shovellng their crappy junk music in mp2.5 format at us?

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BOBCHEWIE
post Jan 3 2010, 13:33
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QUOTE (krabapple @ Dec 1 2009, 23:56) *
This is all peachy for those recording DSD files or tapping into their SACD player DACs, but could one of you gurus please build an SACD ripper for the rest of us? Thanks.



two methods..both are hard work..(1) tapping in binary stream at 88.khz to pcm convert (2) analogue transfer ie re record all the channels FL FR, CEN LFE, LS RS as stereos and recombine (in sync) back into surround (its all done by ear mate and fine tuning) record all 3 stereos front, centre,rear at 92khz 24bit then save out as wma ot wave or 6 mono wavs, then convert to mlp and make dvd audio out of...or oversample at 192khz 24bit all 3 stereos and downsample to 96/24 and yes i did save out a huge 192khz 24bit 5.1 wave file that was HUGE and vlc player played it back with no fuss...also if anyone is interested flac supports 192/24 5.1 channels too and a wee bit small in size than wav...
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rpp3po
post Jan 3 2010, 14:15
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QUOTE (BOBCHEWIE @ Jan 3 2010, 12:14) *
scream, have a hissy fit and take my ball back...why the heck do we have to put up with this 'anti hires' propaganda all the time..


I don't think anybody is interested in your opinion here. Deliver some facts or keep it to yourself. It's the hires propagandists who could not deliver a single, time-synced, level-matched double blind test result where anybody, including Bob Katz, would have been able to differentiate a hires track from a high quality 16bit/44.1kHz mixdown.

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Axon
post Jan 3 2010, 14:22
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QUOTE (BOBCHEWIE @ Jan 3 2010, 05:14) *
QUOTE (rpp3po @ Nov 24 2009, 10:23) *
Raw DSD means saving about 3 GB per 70 minute stereo album. Kind of a waste when you can save 48kHz files, that should perfectly* preserve the 0-22 kHz band, and only need about 475 MB (FLAC). That's over 2.5 GB wasted, not for discarded but inaudible recorded information but plain quantization distortion, added by an inferior form of digitalization. Why save 2.5 GB of garbage per album? Has there ever been just one solid positive blind test for DSD?

* talk about differences of -110db or lower

ok i have idea lets make or convert our audio to 11khz mono and knock it into mp3 format..after all, its about quantity over quality isnt it?
the next time i read an article that tells me the audio customers arent interested in quality and the SACD /DVD Audio format is dead ..i will
scream, have a hissy fit and take my ball back...why the heck do we have to put up with this 'anti hires' propaganda all the time..what ?so that el cheapo music company can keep on shovellng their crappy junk music in mp2.5 format at us?

I don't think "quality" means what you think it means.

If one cannot tell apart SACD from MP3 in a blind test, then as far as the parameters of that test are concerned, that does not mean that choosing MP3 means sacrificing quality. It means that SACD is equal in quality to MP3. It does mean sacrificing other things, but audio quality is not necessarily among them.
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Martel
post Jan 4 2010, 09:53
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QUOTE (BOBCHEWIE @ Jan 3 2010, 12:14) *
...why the heck do we have to put up with this 'anti hires' propaganda all the time...

QUOTE (Wikipedia)
Propaganda is a form of communication aimed at influencing the attitude of a community toward some cause or position. As opposed to impartially providing information, propaganda in its most basic sense, presents information primarily to influence an audience. Propaganda often presents facts selectively (thus lying by omission) to encourage a particular synthesis, or uses loaded messages to produce an emotional rather than rational response to the information presented.
Which part of this describes what HA community is doing? Last time I checked, people here demanded (verifiable) evidence to support one's claims. I guess it's the other way around in your case (hi-res propaganda).


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udauda
post Jan 4 2010, 13:39
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I wonder, how come Pro-DSD never comes up with a solid study like this:
DVD-A versus SACD
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rpp3po
post Jan 4 2010, 14:31
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Excellent paper, thanks!
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krabapple
post Jan 8 2010, 22:05
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I remember this one from a few years back....I forget whether the full version was ever published or not; the link to the master's thesis appears dead. Very interesting and mostly thorough work ...

But regarding the four out of 145 subjects who scored significantly better than chance on their first tests:

"All four of these tests occurred within the final four
days of testing. By that point the testing schedule was
full, so unfortunately these individuals could not be
brought back for follow-up verification tests."

Grr. I hate when this happens!

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ElfeJoyeux
post Mar 11 2010, 22:34
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Hello,

I just discovered this awesome topic during some googling. Let me explain : I bought a hybrid SACD. I can listen to the CD dayer without any problem. I recently noticed that the disk is damaged, as shown on this picture. So before it's totally broken, I'd really like to extract the SACD part (for the day I'll buy a 5.1 sound system). I'm really annoyed because it's a very rare disk (only sold during 2 weeks because the band wasn't satisfied by the songs... so it's now impossible to buy this disk).

Thanks to SebastianG, I know I can convert the extracted datas to PCM (then FLAC or anything else).

The question is now : how to extract the datas. I have no SACD player, and know no one who owns one. And I don't want to buy one just for a single use...
- is there a way to use an old PS3 for this ?
- is there here someone who successfully extracted a SACD datas, and who is kind enough to do this for my disk ? (I can send it worldwide and pay for you to send it back to me smile.gif )

Thanks !
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Cyril
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Hobbit13
post Jul 19 2010, 10:48
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QUOTE (ElfeJoyeux @ Mar 11 2010, 22:34) *
The question is now : how to extract the datas. I have no SACD player, and know no one who owns one. And I don't want to buy one just for a single use...
- is there a way to use an old PS3 for this ?
- is there here someone who successfully extracted a SACD datas, and who is kind enough to do this for my disk ? (I can send it worldwide and pay for you to send it back to me smile.gif )

Thanks !
--
Cyril



If you have a PS3, and want a 5.1 rip, you can reflash a 1.0 version of the PS3 firmware. If you then play the SACD, the digital audio out will contain an unencrypted DTS stream, which you can easily capture with a decent sound cart with SP/dif in.

For a high-res digital rip, very few options are available, as you would have know from reading this thread.

Cheers
Hobbit13
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Sik_Lescinovid
post Nov 27 2011, 18:28
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Has anyone so far noticed that when creating a 24 bit /352.8 kHz WAV file using
CODE
cat input.dff | dsd2pcm 2 m 24 | sox -t raw -r 352.8k -s -3 -c 2 - output.wav

the output file has a loud click at the very beginning lasting for only about 60 samples or so? I also noticed that there seems to be an issue when trying to convert larger files (say a .dff of the whole WYWH album)
which limits the output to only a clip of about 10 minutes or so (I've checked that the conversion doesn't stop due to lack of space or anything like that).

Additionally I wanted to ask if anyone has done any more work on this converter or if there is any more info on this project apart from this thread and the dsdiff plugin for foobar thread?
I'm interested to find out if converting .dff to .wav using dsd2pcm and then downsampling to 24/96 with Saracon yields a better sound than using Saracon's DSD2PCM module, i.e. if less filtered conversions might sound better
than the heavily filtered ones from Saracon.

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