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Can we play genuine DSD Stream with PCs?, Moved (back) to Audio Hardware by request
Happy Water
post Feb 15 2012, 23:41
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Hi,

now ripping a SACD is possible and we can play SACD ISO with foobar2000.
But still, we must convert DSD stream to a PCM stream and then output it to the PC soundcard or
a DA/C through USB 1394 HDMI or SPIDF, etc..

I saw the autodetect ASIO Driver Mode in foo_input_sacd plugin Preferences page of foobar2000, but I don't know any Devices(both soundcard and DAC)
that supports it. Now I'm really looking forward to a scheme that can play genuine DSD Stream with PC...

Does anyone have any ideas or suggestions?

Thanks...
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Happy Water
post Feb 21 2012, 00:42
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Again,

Now I've just heard about the "Mytek digital stereo 192-dsd dac", which is claimed can support DSD stream playing. But I am not sure whether it can work with foobar2000(or maybe some other secondary equipments are needed), and I am going have one if it can.

Anyone who can express it?

Thanks...

This post has been edited by Happy Water: Feb 21 2012, 01:37
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saratoga
post Feb 21 2012, 01:34
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Internally most DACs seem to prefer to work with DSD by first converting it to the same intermediate format as they do PCM audio, so I'm not sure theres any point in trying to play DSD on the PC. You can get more or less the same effect using the PCM interface as the DSD one (at least assuming you have a high enough sampling rate).
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Happy Water
post Feb 21 2012, 10:58
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QUOTE (saratoga @ Feb 21 2012, 01:34) *
Internally most DACs seem to prefer to work with DSD by first converting it to the same intermediate format as they do PCM audio, so I'm not sure theres any point in trying to play DSD on the PC. You can get more or less the same effect using the PCM interface as the DSD one (at least assuming you have a high enough sampling rate).

"intermediate format" do you mean that some DACs prefer to convert PCM into Analog Signal by using delta-sigma modulation(or just like a Class D Amplifier)?
I'm not sure what do you mean.But in my poor opinion, only a few Products perform such actions, so it doesn't matter a lot to the topic.

In theory, we need just a LPF to convert a DSD bit stream into an Analog Signal(however, to achieve a good sound quality, a good implementation is necessary).
And converting DSD into PCM will loss almost all the advantages of DSD(Base Band SNR, bitrate, etc.). <--Anyway, PCM does not have such a high sampling rate as DSD do, thus, we can never Cover(convert without any theorical losses) a DSD stream by a PCM stream.
That is why I turned to Devices that *REALLY* support DSD playing. Even if there is not so much difference in sound quality that I can hear.
In one word, I'm trying to find out a theorically ideal device(no matter Soundcard or DAC).

On the other hand, just like we rip CDs into PC. PC can provide us flexible interface, large storage space and lots of other conveniences. So it is not pointless in trying to play DSD on the PC.

Still on my way...
Please help.Thanks.

This post has been edited by Happy Water: Feb 21 2012, 11:01
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knutinh
post Feb 21 2012, 11:24
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QUOTE (Happy Water @ Feb 21 2012, 11:58) *
"intermediate format" do you mean that some DACs prefer to convert PCM into Analog Signal by using delta-sigma modulation(or just like a Class D Amplifier)?
I'm not sure what do you mean.But in my poor opinion, only a few Products perform such actions, so it doesn't matter a lot to the topic.

I think you will find that nearly all available products do just this. And probably exactly all of the good ones.
QUOTE
And converting DSD into PCM will loss almost all the advantages of DSD(Base Band SNR, bitrate, etc.).

DSD have to my knowledge no advantages over PCM.
QUOTE
<--Anyway, PCM does not have such a high sampling rate as DSD do, thus, we can never Cover(convert without any theorical losses) a DSD stream by a PCM stream.

If you listen to the music, in principle any non-trivial volume adjustement (analog or digital, in the studio or at home) introduce errors.

Perhaps you should ask yourself "how do I get the best experiences while listening to music?", instead of "what theoretical benefit can I get from doing this?". My suggested reply to my suggested question would be: purchase the very best releases of the music that you like, and the playback hardware needed to play whatever that media (or convert it to something of your choice). Spend some time and money on choice and placement of loudspeakers and room.

-k
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Arnold B. Kruege...
post Feb 21 2012, 14:01
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QUOTE (Happy Water @ Feb 21 2012, 04:58) *
And converting DSD into PCM will loss almost all the advantages of DSD(Base Band SNR, bitrate, etc.).


AFAIK none of those purported advantages actually exist.

For example both DSD and PCM can provide improved Base Band dynamic range.

You seem to be confused about information theory which states that no matter how you do it, there are iron rules that connect data rates, bandwidth, and dynamic range.

QUOTE
<--Anyway, PCM does not have such a high sampling rate as DSD do,


Again you seem to be making up science as you go along. Sampling bits and sampling words are not directly analogous.

I repeat without fear of effective contradiction that Information Theory still rules.

It is pretty well known among the more technically sophisticated members of the audio industry that DSD was invented by Sony for the purposes of IP protection, not better sound.



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Happy Water
post Feb 21 2012, 22:18
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Thanks for replies...

Actually, I am a senior student of Information Engineering. As a senior, I made those conclusions with confidence, but now it seems I am still quite young.

QUOTE (Arnold B. Krueger @ Feb 21 2012, 14:01) *
It is pretty well known among the more technically sophisticated members of the audio industry that DSD was invented by Sony for the purposes of IP protection, not better sound.


Now I think this should be the key word for me as I do originally thought that Sony invented DSD for better sound.

However, there is still one question confuses me.
How much loss is there when we convert DSD to PCM(especially 176.4KHz 24bit, by foo_input_sacd Plugin or AudioGate) for playing those Albums which are claimed recorded and mixed under Pure DSD Environment? (if any?)

BTW, I heard that only a few Music Albums are recorded under pure DSD environment while the others are converted from PCM tracks.
Is it true?
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Wombat
post Feb 21 2012, 22:36
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QUOTE (Happy Water @ Feb 21 2012, 22:18) *
Now I think this should be the key word for me as I do originally thought that Sony invented DSD for better sound.

However, there is still one question confuses me.
How much loss is there when we convert DSD to PCM(especially 176.4KHz 24bit, by foo_input_sacd Plugin or AudioGate) for playing those Albums which are claimed recorded and mixed under Pure DSD Environment? (if any?)

BTW, I heard that only a few Music Albums are recorded under pure DSD environment while the others are converted from PCM tracks.
Is it true?

The chances there is a loss isnīt that high. The DSD files they offer may have been PCM before while they were mastered. Software programs like Pyramix advertise to use 24bit/354.2kHz PCM as intermediate format for processing. So you see you can pretty nicely create the complete DSD signal in PCM, even with all the noise smile.gif
It doesnīt work the other way around.
When converting 1bit DSD to PCM 20bit and 48kHz sampling rate may be already enough. At 24kHz the noise in DSD is already ~60dB higher as with PCM, no music up there you will miss imho.
Just realize how they claim the advantage to have no ringing filter up there with DSD. Having some pre-ringing at 24kHz makes your music sound bad but 60dB more noise up there doesnīt. Silly, isnt it?
To be on the safe side you may use 88.2kHz but everything above only wastes space with recreating huge amounts of noise that in reality better should be filtered out.
We have some people here with more insight here, just my thoughts.

Edit: And about PURE DSD production you should wonder who still uses 1-bit DSD ADCs when multi-bit DSD became the norm. So the data they sell you as 1-bit DSD may not be so PURE

This post has been edited by Wombat: Feb 21 2012, 22:41
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saratoga
post Feb 21 2012, 23:31
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QUOTE (Happy Water @ Feb 21 2012, 04:58) *
QUOTE (saratoga @ Feb 21 2012, 01:34) *
Internally most DACs seem to prefer to work with DSD by first converting it to the same intermediate format as they do PCM audio, so I'm not sure theres any point in trying to play DSD on the PC. You can get more or less the same effect using the PCM interface as the DSD one (at least assuming you have a high enough sampling rate).

"intermediate format" do you mean that some DACs prefer to convert PCM into Analog Signal by using delta-sigma modulation(or just like a Class D Amplifier)?
I'm not sure what do you mean.But in my poor opinion, only a few Products perform such actions, so it doesn't matter a lot to the topic.


I think the most common approach is to lowpass/decimate the DSD into PCM and then play that down the same pathway used for normal PCM audio. This way the same chip can be used for either format.

QUOTE (Happy Water @ Feb 21 2012, 04:58) *
How much loss is there when we convert DSD to PCM(especially 176.4KHz 24bit, by foo_input_sacd Plugin or AudioGate) for playing those Albums which are claimed recorded and mixed under Pure DSD Environment? (if any?)


DSD systems tend to lowpass at 50kHz, so compared to that 24/176.4k would have essentially no loss (well baring some tiny rounding error that would be far below the noise floor anyway).
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Wombat
post Feb 21 2012, 23:40
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QUOTE (saratoga @ Feb 21 2012, 23:31) *
I think the most common approach is to lowpass/decimate the DSD into PCM and then play that down the same pathway used for normal PCM audio. This way the same chip can be used for either format.

My AK4396 is a delta-sigma type that should be very common. It can play PCM and DSD. It takes PCM and converts it to its own "Bitsream format" I guess it is multi-bit DSD. So there must be several topologies out there.
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Roseval
post Feb 22 2012, 10:59
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QUOTE (saratoga @ Feb 21 2012, 23:31) *
DSD systems tend to lowpass at 50kHz


This is recommanded by the DSD standard.
Probably to protect your system against the high amount of quantization noise.


DSD quantisation noise with a sampling rate of 64 times 44.06 kHz.
Malcolm Hawksford, Essex University, UK

A good read:
Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications
Stanley P. Lipshitz and John Vanderkooy
http://sjeng.org/ftp/SACD.pdf


--------------------
TheWellTemperedComputer.com
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Arnold B. Kruege...
post Feb 24 2012, 14:01
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QUOTE (Happy Water @ Feb 21 2012, 16:18) *
How much loss is there when we convert DSD to PCM(especially 176.4KHz 24bit, by foo_input_sacd Plugin or AudioGate) for playing those Albums which are claimed recorded and mixed under Pure DSD Environment? (if any?)


Very little to vanishing, depending on how the recording was sourced. I guess that DSD claims HF band pass to 100 KHz, while 24/192 has 4 KHz less band pass, only 96 KHz. ;-) Trust me, not even the most enthusiastic technically credible advocates of high sample rates would argue that any veils at all are lifted by increasing HF band pass from 96 KHz to 100 KHz. I'm sure that there are some from the lunatic fringe who would fight to the death for those 4 KHz above 96 KHz, but they are lunatics, right? ;-) So, this minuscule HF band pass difference is a draw in the minds of everyone with a mind.

That leaves us with dynamic range. 24/192 has a theoretical dynamic range of 144 KHz from full use of the 24 bits with a low uniform low noise floor up to 96 KHz. DSD has a higher noise floor below 20 KHz, which most authorities put in the range of about 120 dB. Above 20 KHz or so, the noise floor is allowed to rise significantly to as little as 50 or 60 dB. Now from a listening standpoint this is all very good performance, and gross overkill.

The big idea you need to understand is that from a theoretical dynamic range standpoint, 24 bit PCM walks all over DSD. If you transcribe a DVD recording to 192/24 there has not been any actual theoretical loss due to the 192/24 digital format.

The bottom line is that converting a SACD to 24/192 PC has no inherent theoretical losses with any merit, and anybody who says otherwise either does not know the relevant facts or is invoking some kind of voodoo theory of digital sound reproduction. A similar contemporary voodoo theory of digital sound reproduction is the idea that repeated FLAC compression (without a basic format change) causes audible changes or that the simple error-free move of a digital audio file from one area or kind of digital storage to another causes audible changes.

QUOTE
BTW, I heard that only a few Music Albums are recorded under pure DSD environment while the others are converted from PCM tracks.
Is it true?


Apparently all too true. An independent consultant who was hired by an online retailer of higher resolution recordings to transcribe DVD-A and SACD recordings reports that something like half of all the recordings he converted were apparently originally transcribed from appreciably lower resolution sources such as legacy digital files with 16 bit data and 48 KHz sample rates, or analog tape with even worse dynamic range and high frequency extension than the legacy digital recordings. Other reliable sources have reported similar findings based on analysis of smaller but representative samples of commercial recordings.

In their rush to build up a large catalog and profit again from reselling recordings that had already been sold to zillions of consumers once or twice before, the commercial advocates of so called high resolution recordings ended up cooking their own gooses! In my view they were just supporting what many of us already knew, which is that the so-called high resolution formats had nothing more in terms of listening enjoyment to offer to music lovers, other than perhaps a quick remastering job by various people, some unknown with questionable talent, experience, and credentials.

Interestingly enough it appears that there were few if any reviewers who listened to these so-called hi-res recordings and reported any differences among recordings depending whether they did or did not come from bandpass-limited sources.

Furthermore, the dynamic range of musical recordings is limited by other factors than the recording equipment, and in fact diligent searching finds none that actually tax the capabilities of the Redbook CD forma (16 bits). Furthermore there are just a tiny handful of microphones and loudspeakers that have anything like 50 KHz bandpass, let alone 100 KHz. Such response as they may have is generally limited to only sources and listeners who sit almost perfectly on-axis.

This post has been edited by Arnold B. Krueger: Feb 24 2012, 14:03
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