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DSD-2-PCM -- proof of concept, test sample and source code here
Axon
post Nov 24 2009, 21:56
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Look into the dynamic range of SACD at 22khz sometime. You'll be shocked.
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rpp3po
post Nov 25 2009, 03:00
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QUOTE (LukeS @ Nov 24 2009, 21:39) *
48kHz files do not preserve 0-22kHz band perfectly, the brick wall effect means that the high pass filters have to cut into much lower frequencies for seamless playback.


I think you'll have a hard time explaining this, because it's simply not true. What's the "brick wall effect"? Do you have that from Stereophile magazine?
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LukeS
post Nov 25 2009, 04:20
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QUOTE (rpp3po @ Nov 24 2009, 21:00) *
QUOTE (LukeS @ Nov 24 2009, 21:39) *
48kHz files do not preserve 0-22kHz band perfectly, the brick wall effect means that the high pass filters have to cut into much lower frequencies for seamless playback.


I think you'll have a hard time explaining this, because it's simply not true. What's the "brick wall effect"? Do you have that from Stereophile magazine?


This is not from some magazine; I don't read those audiophile \ home audio magazines. This is basic signal processing, learned it in my EE classes in college and has been reinforced by stuff I have done in practice and what I have read in professional technical publications. You can read about this in any DSP textbook too.

The "brick wall" is an easy way of saying the frequency cutoff due to nyquist frequency which is half the sample rate. You can not accurately capture frequencies above the nyquist frequency. The problem when converting audio into 44.1kHz or 48kHz is the audio you are going to convert can not contain frequencies above 22.05kHz or 44kHz respectively. All data sampled at higher sample rates (which master recordings are) contains higher frequency information then lower sample rate data unless the "brick wall" low pass filter was purposely set below the nyquist frequency.

A low pass filter is required to cutoff frequencies above the nyquist frequency to eliminate picking up aliasing of higher then nyquist frequency, frequency components of the original input waveform. This filter distorts and attenuates high frequencies near the low pass filter cut off point because if the filter was setup to be a sharp low pass filter it would cause auditable high frequency distortions. Whether most people can hear this is arguable but the affect is there and that is just another reason why high fedility formats like 96kHz are nice. DSD also does not suffer from this brick wall low pass filter problem.

It is also worth saying that 48kHz will not suffer as bad as CD 44.1kHz from this effect.

There are references to this all over the place but here is a reference I can remember and another explanation from a AES technical paper I read a a month or so ago on HDCD.

QUOTE
It is now clear that the very sharp cutoff "brick wall" filter required to represent an audio signal in digital form with a sampling rate only slightly higher than twice the audible frequency range does cause problems. This issue will be discussed in more depth below.

..

One can predict that fast cut off low pass filters could be problematic. The sharp removal of higher frequency sidebands creates response ripples from rapidly swept sine waves. These will alter peak energy at basilar membrane sites. In addition, ringing responses near transient events can mix with low level harmonics in the signal to create short beats. [17] One can perform simple experiments to demonstrate that both of these are audible.

The necessity of having a sharp cut-off brick wall anti-alias filter in a digital system in order accommodate a sampling frequency near the minimum necessary for the audio band is bound to create some distortions to sound envelopes. This can be easily demonstrated: If one measures the frequency response of such a low pass filter with a very slow frequency sweep, one gets the classical rectangular envelope. If, on the other hand, one increases the sweep speed, the envelope becomes full of ripples because the faster sweep produces sidebands which fall above the cutoff frequency of the filter. Removing the filter restores the rectangular envelope even for the fast sweep. Listening tests have shown that sweeps with different envelopes but the same spectral content do sound different, although it is very difficult to eliminate all system related sources of confusion in these tests.

Music can contain very complex high frequencies which produce complicated envelopes. It is a logical extension of the above results that if removing high frequencies, which in themselves may not be audible, produces changes in the envelope of audible sounds, then a change in those sounds can be heard. Changes in the characteristics of the filters in the neighborhood of their transition regions can have an effect on the behavior of signal envelopes.


There are other areas of the article that discuss how the brick wall nyquist filter affects the lower frequencies. I didn’t feel like rereading the whole thing again to copy and past each reference.

This post has been edited by LukeS: Nov 25 2009, 04:22
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Axon
post Nov 25 2009, 04:59
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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
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LukeS
post Nov 25 2009, 05:21
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QUOTE (Axon @ Nov 24 2009, 22:59) *
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
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saratoga
post Nov 25 2009, 05:25
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QUOTE (LukeS @ Nov 24 2009, 22:20) *
QUOTE (rpp3po @ Nov 24 2009, 21:00) *
QUOTE (LukeS @ Nov 24 2009, 21:39) *
48kHz files do not preserve 0-22kHz band perfectly, the brick wall effect means that the high pass filters have to cut into much lower frequencies for seamless playback.


I think you'll have a hard time explaining this, because it's simply not true. What's the "brick wall effect"? Do you have that from Stereophile magazine?


This is not from some magazine; I don't read those audiophile \ home audio magazines. This is basic signal processing, learned it in my EE classes in college and has been reinforced by stuff I have done in practice and what I have read in professional technical publications. You can read about this in any DSP textbook too.



Thats a very long way of not answering his question at all. Let me try and focus you:

What's the "brick wall effect"? Why does it mean that "high pass filters have to cut into much lower frequencies for seamless playback"? Although I think you meant lowpass not high pass wink.gif

Edit: To be clear, you're claiming that you can't go up to 22kHz with good alias rejection using a 48kHz sampling rate, or rather that a DAC can't accurately reproduce ~.46*fs, which I think is a little nuts.

This post has been edited by Mike Giacomelli: Nov 25 2009, 05:29
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LukeS
post Nov 25 2009, 05:29
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QUOTE (Mike Giacomelli @ Nov 24 2009, 23:25) *
Thats a very long way of not answering his question at all. Let me try and focus you:

What's the "brick wall effect"? Why does it mean that "high pass filters have to cut into much lower frequencies for seamless playback"? Although I think you meant lowpass not high pass wink.gif


The high pass filter is a typo and after I realized it I was locked out from editing it, apparently there is a edit timeout on this board sad.gif Also I think I explained the brick wall well enough

QUOTE
Edit: To be clear, you're claiming that you can't go up to 22kHz with good alias rejection using a 48kHz sampling rate, or rather that a DAC can't accurately reproduce ~.46*fs, which I think is a little nuts.

I was not referring to the end user DAC \ output side of things, I was referring to input ADC, processing, and down sampling to 48kHz of the original audio data that causes issues. Most people can't here this stuff or care.

This post has been edited by LukeS: Nov 25 2009, 05:47
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saratoga
post Nov 25 2009, 05:55
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QUOTE (LukeS @ Nov 24 2009, 23:29) *
Also I think I explained the brick wall well enough


The reason I ask is that a brick wall filter doesn't "distort and attenuates high frequencies near the low pass filter cut off". A brick wall filter is shaped like a brick. Infinite attenuation on one side, unity on the other. I would not call the use of something not a brickwall the brickwall effect smile.gif

QUOTE (LukeS @ Nov 24 2009, 23:29) *
QUOTE
Edit: To be clear, you're claiming that you can't go up to 22kHz with good alias rejection using a 48kHz sampling rate, or rather that a DAC can't accurately reproduce ~.46*fs, which I think is a little nuts.

I was not referring to the end user DAC \ output side of things I am discussing the input ADC, processing, and down sampling to 48kHz of the original audio data that causes issues. Most people can't here this stuff or care.


Yes, and if the filter in a DAC can do it, then the same filter if used in a resampler can also do it. So clearly it would be (and absolutely is) possible.
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vincefalks
post Nov 25 2009, 08:02
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QUOTE (LukeS @ Nov 24 2009, 22:39) *
I am very new to the SACD format but I am interested in picking up a second hand SACD player that outputs DSD over HDMI and trying to grab the DSD signals before they get encrypted and sent over HDMI, it may offer a less altered signal.
I myself am not the biggest technical expert on the mathematics of PCM or DSD, but I thought nothing can be altered to the signal within the (digital) DSD domain - only at analog/PCM stage of the player, maybe. I might be wrong though. There certainly are other players which do not cut off ultra high freuencies like that Denon.

QUOTE
My plan exactly, I will keep the originals for archival purposes and make a copy that is converted it to a lossless PCM format like FLAC if a good conversion becomes available.

I concur.

QUOTE
Also the storage could be greatly reduced is a loss less container format was made to store the raw DSD channels.
Well THAT sir, is DST - it's like MLP for PCM/DVD-A. Currently, only rare pro applications can encode DST, but I hope in future we will have acces to such a facility.
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knutinh
post Nov 25 2009, 08:36
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QUOTE (LukeS @ Nov 24 2009, 21:39) *
Also the storage could be greatly reduced is a loss less container format was made to store the raw DSD channels.


I think that LPCM offers lossless, uncompressed output from available, high-quality ADCs.

FLAC offers lossless compression of LPCM.

DSD offers unprocessed storage*) of ADCs that are no longer viable. For all other cases, it is simply a bad trade-off between filesize and quality.

DST offers lossless compression of DSD.

http://en.wikipedia.org/wiki/Sacd#DST

-k

*)This is a benefit invented by PR and poorly educated audiophiles.
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LukeS
post Nov 25 2009, 09:16
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QUOTE (vincefalks @ Nov 25 2009, 02:02) *
QUOTE (LukeS @ Nov 24 2009, 22:39) *
I am very new to the SACD format but I am interested in picking up a second hand SACD player that outputs DSD over HDMI and trying to grab the DSD signals before they get encrypted and sent over HDMI, it may offer a less altered signal.
I myself am not the biggest technical expert on the mathematics of PCM or DSD, but I thought nothing can be altered to the signal within the (digital) DSD domain - only at analog/PCM stage of the player, maybe. I might be wrong though. There certainly are other players which do not cut off ultra high freuencies like that Denon.

See the attached a screen shot from the Denon DVD-955 / DVD-2910 manual, there is some filtering going on in the player. Either this is an option that is set on the DACs (most likely I would think), in the analog circuity, or the Denon player post-processes the DSD signal and applies a 50kHz or 100kHz filter before sending it to the DACs. If this is the case it is probably done in the sony DSD chip but it would seem to be way to processing intensive to do so and is probably done in the analog domain. I have some digging to do in the DAC datasheet to find out exactly.

QUOTE
QUOTE
Also the storage could be greatly reduced is a loss less container format was made to store the raw DSD channels.
Well THAT sir, is DST - it's like MLP for PCM/DVD-A. Currently, only rare pro applications can encode DST, but I hope in future we will have acces to such a facility.

Haha, yea that's the problem, I don't have thousands of dollars or access to pro tools sad.gif

QUOTE (knutinh @ Nov 25 2009, 02:36) *
QUOTE (LukeS @ Nov 24 2009, 21:39) *
Also the storage could be greatly reduced is a loss less container format was made to store the raw DSD channels.


I think that LPCM offers lossless, uncompressed output from available, high-quality ADCs.

FLAC offers lossless compression of LPCM.

DSD offers unprocessed storage*) of ADCs that are no longer viable. For all other cases, it is simply a bad trade-off between filesize and quality.

DST offers lossless compression of DSD.

http://en.wikipedia.org/wiki/Sacd#DST

-k

*)This is a benefit invented by PR and poorly educated audiophiles.


DST would work but I have not found a no low cost or free software to convert raw DSD to DST or to combine the 6 mono DSD files captured and compress them in a DST file. There is no mainstream information no how to rip the DST \ DSF files on the SACD so that option is out.

FLAC and others are PCM formats and I would like to compress the raw DSD files I capture so those formats are out. ZIP works but it is not optimized for DSD, it will do until I can do something with the DSD files anyways. If a free or low cost tool comes about to convert DSD to PCM or if I figure out why the program SebastianG is not liking my DSD files I will store the converted PCM output in FLAC. I may just end up buying a player that converts the DSD to high sample rate LPCM, grab the unencrypted signals in the player, save that to FLAC, and be done with it.

This post has been edited by LukeS: Nov 25 2009, 09:49
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Martel
post Nov 25 2009, 10:24
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QUOTE (LukeS @ Nov 25 2009, 05:21) *
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. smile.gif


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IE4 Rockbox Clip+ AAC@192; HD 668B/HD 518 Xonar DX FB2k FLAC;
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SebastianG
post Nov 25 2009, 10:32
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QUOTE (LukeS @ Nov 25 2009, 09:16) *
See the attached a screen shot from the Denon DVD-955 / DVD-2910 manual, there is some filtering going on in the player.

Of course there is. At 20 kHz you still have a dynamic range of 110 dB, but it goes down very quickly. Starting at 100kHz you have pure noise. The quantization noise is so powerful up there that is it probably 10^4 times larger than anything "natural" you would want to record. So, of course you need filtering. Anything above 100kHz should be attenuated by at least 100 dB, starting the roll off at 50kHz or so. You can beat this easily with a simple PCM system running at 200 kHz and 8 bits/sample at nearly half the data rate of DSD64. Go figure.

QUOTE
I think that LPCM offers lossless, uncompressed output from available, high-quality ADCs.

IMHO, the terms "lossless" and "lossy" only apply to a process that turns one signal/bitstream to another.

Cheers,
SG
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rpp3po
post Nov 25 2009, 12:04
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Sorry, if I didn't express myself clearly enough. I did not actually want to ask for an explanation. I thought it was self-explanatory that saying "0-22kHz" made it clear, that - at 48 kHz - 22-24kHz can be used to dump the trash. It is far above anything you could ever hear, if you aren't a bat. It is even high enough to allow some aliasing at the upper end and trade it in for less* ringing in the pass-band.

* up to absolutely inaudible for humans

This post has been edited by rpp3po: Nov 25 2009, 12:12
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2Bdecided
post Nov 25 2009, 12:13
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QUOTE (Axon @ Nov 25 2009, 03:59) *
Sorry for being a bastard
No, carry on.

The problem with forums seems to be that people with zero knowledge and experience (not to mention a mere ten posts) don't know that some of the people they're talking to have actually built the very processes and systems they're misdescribing*.

There's a limit to how much patience you can have.


In the physical world, someone straight out of school wouldn't walk into IBM or Cambridge and start telling the project managers or professors how DSP works. But the equivalent happens in the virtual world all the time.


I'm thinking of writing a polite stock paragraph to paste in on such occasions - it seems the phrase "teaching your grandmother to suck eggs" doesn't translate, or is not well known - and if you have to explain it further, it just causes bad feeling on all sides.

Cheers,
David.

* - which is quite poor in this case, since the thread starts with SebG's java implementation of the same, and the FAQ includes threads which discuss this topic to death.

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knutinh
post Nov 25 2009, 18:02
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QUOTE (LukeS @ Nov 25 2009, 09:16) *
DST would work but I have not found a no low cost or free software to convert raw DSD to DST or to combine the 6 mono DSD files captured and compress them in a DST file. There is no mainstream information no how to rip the DST \ DSF files on the SACD so that option is out.

FLAC and others are PCM formats and I would like to compress the raw DSD files I capture so those formats are out. ZIP works but it is not optimized for DSD, it will do until I can do something with the DSD files anyways. If a free or low cost tool comes about to convert DSD to PCM or if I figure out why the program SebastianG is not liking my DSD files I will store the converted PCM output in FLAC. I may just end up buying a player that converts the DSD to high sample rate LPCM, grab the unencrypted signals in the player, save that to FLAC, and be done with it.

Am I right that you bought/built a grabber for SACD, grabbing the DSD stream prior to the DAC? That seems like an expensive and/or time-consuming hobby project. Surely a few hard-drives or DVD-R disks is no big issue then?

I dont think you can buy an off-the-shelf player that plays SACD, converting it to LPCM and serving it on a non-encrypted channel?

-k
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Axon
post Nov 25 2009, 19:38
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It's a surprisingly well known mod, actually. It's not the first time I've heard of it.
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rpp3po
post Nov 25 2009, 20:38
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I find it interesting, nevertheless. What equipment do you use to convert the electrical signal into a logical bitstream after tapping the wire?
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simmconn
post Nov 26 2009, 01:31
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QUOTE
DST would work but I have not found a no low cost or free software to convert raw DSD to DST or to combine the 6 mono DSD files captured and compress them in a DST file. There is no mainstream information no how to rip the DST \ DSF files on the SACD so that option is out.

FLAC and others are PCM formats and I would like to compress the raw DSD files I capture so those formats are out. ZIP works but it is not optimized for DSD, it will do until I can do something with the DSD files anyways. If a free or low cost tool comes about to convert DSD to PCM or if I figure out why the program SebastianG is not liking my DSD files I will store the converted PCM output in FLAC. I may just end up buying a player that converts the DSD to high sample rate LPCM, grab the unencrypted signals in the player, save that to FLAC, and be done with it.


Time for some Googling. DST is part of IEC14496-5 and the reference code can be obtained at no charge.

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udauda
post Nov 26 2009, 02:31
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QUOTE (Axon @ Nov 24 2009, 19:59) *
You've read Lipshitz/Vanderkooy, right?

The question I am about to ask might be quite irrelevant from the topic but I've been wondering for too long.. tongue.gif

Does the Lipshitz/Vanderkooy paper correspond with this article at ESP?
http://sound.westhost.com/cd-sacd-dvda.htm

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vincefalks
post Nov 26 2009, 03:38
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QUOTE (simmconn @ Nov 26 2009, 02:31) *
QUOTE
DST would work but I have not found a no low cost or free software to convert raw DSD to DST or to combine the 6 mono DSD files captured and compress them in a DST file. There is no mainstream information no how to rip the DST \ DSF files on the SACD so that option is out.

FLAC and others are PCM formats and I would like to compress the raw DSD files I capture so those formats are out. ZIP works but it is not optimized for DSD, it will do until I can do something with the DSD files anyways. If a free or low cost tool comes about to convert DSD to PCM or if I figure out why the program SebastianG is not liking my DSD files I will store the converted PCM output in FLAC. I may just end up buying a player that converts the DSD to high sample rate LPCM, grab the unencrypted signals in the player, save that to FLAC, and be done with it.


Time for some Googling. DST is part of IEC14496-5 and the reference code can be obtained at no charge.

What does this mean? code that you can make an encoder from? Never heard of this concept before. Does MLP have a "reference code", as an example? I thought it was proprietary.
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LukeS
post Nov 26 2009, 07:27
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Wow, I really am messing thing up lately blush.gif ; I got the SACD, DSD ripping working. There was an error in the software program I wrote that translates the data captured from the hardware and puts it into a DSD file. Attached is 28 seconds of the front left channel of Ryan Adams - Gold - 01 - New York, New York. SebastianG program works great.

This post has been edited by LukeS: Nov 26 2009, 07:29
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simmconn
post Nov 26 2009, 08:16
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And I just spent one and a half hours to encode an 8 minutes 44 seconds stereo DSD into DST, only to find that it cannot be played in real time with the foobar2000 DSD plugin.

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2Bdecided
post Nov 26 2009, 11:46
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QUOTE (udauda @ Nov 26 2009, 01:31) *
Does the Lipshitz/Vanderkooy paper correspond with this article at ESP?
http://sound.westhost.com/cd-sacd-dvda.htm
Not really, they're separate and independent, by different authors.

Lipshitz and Vanderkooy show inherent distortion in 1-bit systems. Your links discusses inherent noise. They're two different problems.

Cheers,
David.
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vincefalks
post Nov 26 2009, 12:02
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QUOTE (2Bdecided @ Nov 26 2009, 12:46) *
Lip*****z and Vanderkooy show inherent distortion in 1-bit systems.

This is definitely off-topic, but can I briefly ask - is the distortion in 1-bit audio similar to the distortion in vinyl? And is this 1-bit distortion audible...like vinyl's...thx
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