IPB

Welcome Guest ( Log In | Register )

> Upload forum rules

- No over 30 sec clips of copyrighted music. Cite properly and never more than necessary for the discussion.


- No copyrighted software without permission.


- Click here for complete Hydrogenaudio Terms of Service

5 Pages V  < 1 2 3 4 5 >  
Reply to this topicStart new topic
DSD-2-PCM -- proof of concept, test sample and source code here
rpp3po
post Nov 26 2009, 12:37
Post #51





Group: Developer
Posts: 1126
Joined: 11-February 03
From: Germany
Member No.: 4961



Yawn! pinch.gif Am I really the last one to get where this guy is coming from?

No, of course it not similar! Why should the distortion of 1-bit Delta-Sigma-Modulation correlate with the molecular properties of vinyl and dust, and the mechanical properties of manual tracking and needle movement? That question seems to come from a mind desperately wanting to finally give some sense to a senseless belief system, fed by marketing and idiot group think.

The only similarity is that both distortions are non-linear, but also are vocoders and wah-wah effects.

This post has been edited by rpp3po: Nov 26 2009, 13:10
Go to the top of the page
+Quote Post
vincefalks
post Nov 26 2009, 13:10
Post #52





Group: Members
Posts: 37
Joined: 23-August 09
Member No.: 72571



QUOTE (rpp3po @ Nov 26 2009, 13:37) *
Yawn! pinch.gif Am I really the last one to get where this guy is coming from?

No, of course not! Why should the distortion of 1-bit Delta-Sigma-Modulation correlate with the molecular properties of vinyl and dust, and the mechanical properties of manual tracking and needle movement? That question seems to come from a mind desperately wanting to finally give some sense to a senseless belief system, fed by marketing and idiot group think.

The only similarity is that both distortions are non-linear, but also are vocoders and wah-wah effects.

You're so used to wild audiophiles claiming crazy theories that you automatically assume someone bringing up a question like mine means they also have a wild theory.

How wrong you are!

THIS is where I am coming from: Very uneducated vinyl zealots (I believe Michael Fremer said this on a youtube video) think that vinyl's warmth is what "lifelike analog" sound is (compared to "cold" jagged digital). (when in fact it is certain distortions in the medium). SACD zealots also think that SACD has a "lifelike analog" sound, but I know that SACD also has certain distortions in it.

Thus I've been wondering, PONDERING (and not even claiming!) if there might be a correlation between these two scenarios, or if it's just a funny coincidence. I don't know much about these distortions, and wanted to briefly ask someone who does. this is an audio forum about HELPING others and sharing scientifically valid information. I asked a question, that's all..

A simple no would suffice, along with some explanation and without the attitude and assumptions unrelated to the content of the question. And you didn't answer my second part of the question, "And is this 1-bit distortion audible?".
Go to the top of the page
+Quote Post
rpp3po
post Nov 26 2009, 13:25
Post #53





Group: Developer
Posts: 1126
Joined: 11-February 03
From: Germany
Member No.: 4961



At least you seem believe the life-like-analog-sound-myth of SACD.

When properly dithered, in this context that means only partially dithered, SACD's distortions are inaudible in the 0-22kHz band (and above anyway).
Go to the top of the page
+Quote Post
vincefalks
post Nov 26 2009, 13:40
Post #54





Group: Members
Posts: 37
Joined: 23-August 09
Member No.: 72571



QUOTE (rpp3po @ Nov 26 2009, 14:25) *
At least you seem believe the life-like-analog-sound-myth of SACD.

LOL. Just stop. Nothing I have written has indicated what you just claimed, in fact, the exact opposite. I was comparing both myths to each other and wondering about a possible reason behind SACD's case, of which I don't know much about the science.

QUOTE
When properly dithered, in this context that means only partially dithered, SACD's distortions are inaudible in the 0-22kHz band (and above anyway).

Ah now that's better. t h a n k y o u.

back on topic now...
Go to the top of the page
+Quote Post
2Bdecided
post Nov 26 2009, 15:57
Post #55


ReplayGain developer


Group: Developer
Posts: 4945
Joined: 5-November 01
From: Yorkshire, UK
Member No.: 409



Yes, the distortions of SACD are inaudible. Especially within the 0-16kHz region they're way way way way down. At least at first pass. If people take DSD and process it, things could go very wrong (hence its conversion to multi-bit for processing, discussed earlier in the thread).

If DSD/SACD has a "sound", it's the interaction of all that high frequency noise with every other piece of equipment in the audio chain.

This could just be another myth - I've never seen anyone ABX it - but it's "common knowledge" with many audio engineers I've spoken to (going way back before SACD), that if you pump lots of ultrasonic stuff through an audio system, you change its sound. The pre-dates the modern (~decade-old) "idea" that ultrasonic information is somehow audio - this is plain and simple "all the ultrasonic junk changes the way imperfect audio equipment works".

One effect is certainly real, measurable, and perceptible: the listening room warms up in the vicinity of the audio equipment as all the amps and tweeters work so much harder and generate more heat!

Cheers,
David.
Go to the top of the page
+Quote Post
Canar
post Nov 26 2009, 16:22
Post #56





Group: Super Moderator
Posts: 3327
Joined: 26-July 02
From: princegeorge.ca
Member No.: 2796



So that's what they mean by "warm" sound! Thanks for the clarification, David. cool.gif


--------------------
∑:<
Go to the top of the page
+Quote Post
Axon
post Nov 26 2009, 21:14
Post #57





Group: Members (Donating)
Posts: 1984
Joined: 4-January 04
From: Austin, TX
Member No.: 10933



If SACD really did have a similar distortion profile to vinyl, after spending lots and lots of time on the latter subject, I'm here to you that the distortion on SACD would be composed of lots and lots of jitter.

Which nobody really sees, but if we did, it would kinda pop a bubble in the SACD-superiority thing, wouldn't it? (I'm glossing over some details about low order distortion, but really, you have to be splitting some fine hairs to make that distinction here.)
Go to the top of the page
+Quote Post
SebastianG
post Nov 27 2009, 03:26
Post #58





Group: Developer
Posts: 1317
Joined: 20-March 04
From: Göttingen (DE)
Member No.: 12875



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
Go to the top of the page
+Quote Post
vincefalks
post Nov 27 2009, 03:44
Post #59





Group: Members
Posts: 37
Joined: 23-August 09
Member No.: 72571



QUOTE (SebastianG @ Nov 27 2009, 03:26) *
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 smile.gif. 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.

QUOTE
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.
Go to the top of the page
+Quote Post
SebastianG
post Nov 27 2009, 04:14
Post #60





Group: Developer
Posts: 1317
Joined: 20-March 04
From: Göttingen (DE)
Member No.: 12875



QUOTE (vincefalks @ Nov 27 2009, 03:44) *
That's awesome, SG! I'm going to try a multichannel DFF...super smile.gif. 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.

QUOTE (vincefalks @ Nov 27 2009, 03:44) *
It's here.

Sweet. smile.gif

Cheers,
SG
Attached File(s)
Attached File  dsd2pcm_32bit_x86_linux.zip ( 8.4K ) Number of downloads: 187
Attached File  dsd2pcm_32bit_x86_linux_static.zip ( 412.24K ) Number of downloads: 189
 
Go to the top of the page
+Quote Post
rpp3po
post Nov 27 2009, 04:16
Post #61





Group: Developer
Posts: 1126
Joined: 11-February 03
From: Germany
Member No.: 4961



QUOTE (vincefalks @ Nov 27 2009, 03:44) *
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 sick.gif pain? You yourself might not "have tonnes of info on it", but apparently tonnes of bullshit 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.

Edit: Vincefalks, I've skimmed briefly over what you have actually written so far in this thread. I must admit you soft-pedaled much more than I remembered and guessed from your last comment, which I considered a little insolent as a first reaction to the posted programming work. Especially when it's totally unfounded.

QUOTE (SebastianG @ Nov 27 2009, 03:26) *
Just for kicks and giggles I rewrote most of "dsd2pcm" in C and C++.

Although I don't have any current use for your binary, your code is always an interesting read. That's pretty good work for such a short time! And the code even includes documentation... blink.gif

This post has been edited by rpp3po: Nov 27 2009, 04:53
Go to the top of the page
+Quote Post
vincefalks
post Nov 27 2009, 05:11
Post #62





Group: Members
Posts: 37
Joined: 23-August 09
Member No.: 72571



QUOTE (SebastianG @ Nov 27 2009, 04:14) *
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 wink.gif wink.gif

QUOTE (rpp3po @ Nov 27 2009, 04:16) *
QUOTE (vincefalks @ Nov 27 2009, 03:44) *
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 sick.gif 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!!!


QUOTE
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... cool.gif
Go to the top of the page
+Quote Post
LukeS
post Nov 27 2009, 07:33
Post #63





Group: Members
Posts: 12
Joined: 7-September 09
Member No.: 72985



QUOTE (SebastianG @ Nov 26 2009, 21:26) *
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.

Awesome work biggrin.gif


QUOTE (vincefalks @ Nov 26 2009, 21:44) *
That's awesome, SG! I'm going to try a multichannel DFF...super smile.gif. Oh no hang on - bah I'm a n00b - how do you compile it in windows, or can you provide a windows binary?


Attached is the program compiled for windows in Dev-C++ (bloodshed) GCC compiler.

Windows CMD does not like the piping though, maybe I am using a wrong command?

CODE
type in.dsd | dsd2pcm 1 M 24 > test.pcm
The process tried to write to a nonexistent pipe.


CODE
type in.dsd | dsd2pcm 1 M 24 | "C:\cmdapps\sox\sox.exe" -t raw -r 352800 -s -3 -c 2 - -r 88200 -b 24 test.wav
The process tried to write to a nonexistent pipe.


Not a big deal to use ubuntu to run this
Attached File(s)
Attached File  dsd2pcm.zip ( 140.02K ) Number of downloads: 214
 
Go to the top of the page
+Quote Post
Kees de Visser
post Nov 27 2009, 09:02
Post #64





Group: Members
Posts: 613
Joined: 22-May 05
From: France
Member No.: 22220



There seems to be a growing (niche) market for online sales of hi-res audio. Using an SACD to rip the DSD audio and convert to a pcm format of choice would probably be much easier than sending the hi-res masters from the record company to the online shop.
The problem however I have with this scenario is that the original recording format is not always specified on the SACD. When a 24/44.1 recording e.g. is being released on SACD, ripped for online sales and offered as 24/96 at a premium price, this somehow doesn't seem honest business. Having the surround version available in pcm format without the need to own an SACD player is a good thing though.
Go to the top of the page
+Quote Post
2Bdecided
post Nov 27 2009, 10:59
Post #65


ReplayGain developer


Group: Developer
Posts: 4945
Joined: 5-November 01
From: Yorkshire, UK
Member No.: 409



QUOTE (vincefalks @ Nov 27 2009, 04:11) *
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 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.


DXD is 8fs 24-bit. Assuming it has some kind of anti-alias filter (it clearly does), then it's just PCM generated from an oversampled ADC with some noise shaping.

When they say "PCM-like", what they mean (but would never dream of saying) is that "perfect" PCM at that rate could have a 176.4kHz bandwidth, and 144dB SNR across that entire bandwidth - whereas DXD has:
1. a gentler filter and hence narrower usable bandwidth (that's fine - that could still be called PCM -and would be a fine way of generating 8fs PCM - no need for brick wall filters when there's so much leeway); and
2. far higher noise - especially at higher frequencies. How much? If you ADC straight to DXD, that depends on the choice of ADC architecture (but it's still PCM at the output). If the DXD is converted from DSD, then the noise depends on the original DSD "encoding", and the conversion parameters. It's still PCM at the output - but it's really messy noisy PCM in this case - if you looked at the waveform (for example) there would be plenty of the DSD noise still intact within the DXD signal - so even in silent parts the waveform amplitude would never drop below -60dB - despite DXD allowing it to drop to -144dB (with dither) or -infinity dB (i.e. a stream of digital zeros without dither).

So in essence PCM-like means "like PCM, but with far more high frequency noise".

So, bad PCM really!

It doesn't really matter. What they've done, ten years too late, is catch up with good converter design. When they designed DSD, the idea was to take the output of the best converter available at that time. Unfortunately, by the time they launched DSD, 1-bit converters weren't the "best" - the best was something like a 5-bit Ring DAC or ADC. Now DXD can hold the data from such a converter with very little processing or theoretical degradation.

But as far as human ears are concerned, the DVD-A formats already did that. Most here would claim the CD formats already did that too, except in some extreme and almost irrelevant circumstances.

Cheers,
David.
Go to the top of the page
+Quote Post
smack
post Nov 27 2009, 11:00
Post #66





Group: Members
Posts: 182
Joined: 16-January 02
Member No.: 1046



QUOTE (LukeS @ Nov 27 2009, 07:33) *
Windows CMD does not like the piping though, maybe I am using a wrong command?

Windows does a LF <-> CR/LF conversion on the stdio streams. You need to modify the program and set std::cin and std::cout to binary mode in order to prevent the data corruption caused by that conversion.

At the moment I can't provide you with the exact C++ code for this patch, sorry. Try an internet search, it's a very common Windows programming problem...

This post has been edited by smack: Nov 27 2009, 11:01
Go to the top of the page
+Quote Post
rpp3po
post Nov 27 2009, 12:55
Post #67





Group: Developer
Posts: 1126
Joined: 11-February 03
From: Germany
Member No.: 4961



QUOTE (vincefalks @ Nov 27 2009, 05:11) *
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.


PCM just means pulse code modulation. An analog waveform's amplitude is captured as discrete symbols at regular intervals. But "PCM" doesn't imply any specific form of antialiasing filter! And so it just does not make sense to say something would be different from PCM, because of a "more-relaxed" antialiasing filter!

When you want to tune PCM for a specific kind of application, the selection of a matching antialiasing filter is totally normal. You decide wether you want to max out frequency response (bandwidth) or impulse response within the limits of a chosen sample rate. Either a filter with a steep or gentle slope is chosen accordingly. There are even applications where the filter can be omitted entirely.

The term "like PCM" just refers to common misconceptions in the audio community. Since the world conquering success of the Audio CD, people associate the underlying technology with brick wall filtering. The Audio CD's Nyquist frequency is relatively close to the human-audible frequency range and so steep filters are commonly used to max out available bandwidth.

But different (gentler sloped) filters have always been used, especially for higher sample rate content, even for plain Redbook. That's nothing specific to DXD.

This post has been edited by rpp3po: Nov 27 2009, 13:01
Go to the top of the page
+Quote Post
2Bdecided
post Nov 27 2009, 13:00
Post #68


ReplayGain developer


Group: Developer
Posts: 4945
Joined: 5-November 01
From: Yorkshire, UK
Member No.: 409



QUOTE (rpp3po @ Nov 27 2009, 11:55) *
An analog waveform's amplitude is captured as discrete symbols at regular intervals.
I guess that's why it's "PCM-like" - there's so much noise that a single given discrete symbol (sample!) can have almost no relationship to the original analogue waveform's instantaneous amplitude.

It's a fair distinction they're trying to make - it's just that the terminology is wrong - it is still PCM - just like SBM or noise shaped CDs are still PCM.

I guess an honest description, e.g. "noisy PCM" wouldn't look so good in the marketing! wink.gif

Cheers,
David.
Go to the top of the page
+Quote Post
.halverhahn
post Nov 27 2009, 15:34
Post #69





Group: Members
Posts: 254
Joined: 4-August 03
Member No.: 8168



DXD is PCM - just with a bunch of noise above 48kHz. You pay storage space for saving noise.

I've downloaded from http://www.2l.no/hires/index.html the Divertimenti Stereo DXD (344 MB) and created a spectrogram.

The DXD 24Bit/352.8kHz


The 24Bit/96kHz version (downsampled with Audition from DXD)




--------------------
.halverhahn
Go to the top of the page
+Quote Post
rpp3po
post Nov 27 2009, 17:47
Post #70





Group: Developer
Posts: 1126
Joined: 11-February 03
From: Germany
Member No.: 4961



Wow! The DXD track is certainly nothing Bruce Wayne should playback in his Batcave. The amount and intensity of ultrasonic noise is insane!
Go to the top of the page
+Quote Post
krabapple
post Nov 29 2009, 03:56
Post #71





Group: Members
Posts: 2157
Joined: 18-December 03
Member No.: 10538



QUOTE (vincefalks @ Nov 26 2009, 23:11) *
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).


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.

This post has been edited by krabapple: Nov 29 2009, 04:02
Go to the top of the page
+Quote Post
spoon
post Nov 29 2009, 20:54
Post #72


dBpowerAMP developer


Group: Developer (Donating)
Posts: 2723
Joined: 24-March 02
Member No.: 1615



As the noise is constant this shows the levels quite well:



Especially for all those porpoises out there:

Animal hearing ranges

Anyone know the filter design in SACD players?

This post has been edited by spoon: Nov 29 2009, 20:55


--------------------
Spoon http://www.dbpoweramp.com
Go to the top of the page
+Quote Post
spoon
post Nov 29 2009, 22:51
Post #73


dBpowerAMP developer


Group: Developer (Donating)
Posts: 2723
Joined: 24-March 02
Member No.: 1615



QUOTE
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.


Makes interesting reading:

http://grad.uprm.edu/tesis/latorrenavarro.pdf

QUOTE
In addition, Sony claims to have developed a decimation procedure, named Super Bit Mapping Direct Down conversion, which uses a “super-power one-stage” FIR digital filter/noise shaper with 32,639 taps to down-convert the DSD signal into a CD quality signal in a single stage [1]. This allows for a CD with the highest possible audio quality.

According to manufacturers, DSD can offer a frequency response from DC to 100 kHz and a residual noise power of -120 dB thanks to its 5th-order noise shaping filters [1]. However, given this noise shaping technique, the residual noise power is not constant throughout the signal bandwidth. By using noise-shaping filters of higher order, it is possible to increase the resolution in the audio band at the expense of resolution at exceedingly (inaudible) high frequencies. These results are valid however only for static, non-transient signals. Transient signals will have poor resolution in a one-bit system. If the signal does not endure for a long enough time, the noise shaper of the one-bit system will not minimize the error.

As indicated by its proponents, DSD offers the advantage of eliminating the decimation and interpolation processes necessary in PCM modulation, thus preventing additional-unnecessary re-quantization noise from polluting the signal. In the SACD "Scarlet Book" Sony and Philips mandate the use of a 100 kHz low-pass filter in SACD mastering so that when the playback volume at standard DSD level is equivalent to 100W, the noise component "outside the audible sound spectrum" is 1W or less. In addition, SACD players must low-pass-filter their analog output above 50 kHz.




--------------------
Spoon http://www.dbpoweramp.com
Go to the top of the page
+Quote Post
krabapple
post Nov 30 2009, 06:33
Post #74





Group: Members
Posts: 2157
Joined: 18-December 03
Member No.: 10538



QUOTE (spoon @ Nov 29 2009, 16:51) *
QUOTE
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.


Makes interesting reading:

http://grad.uprm.edu/tesis/latorrenavarro.pdf


Interesting listening tests, though also a bit curious. Not sure how someone managed to get 81.5% correct in a 16-trial listening test of 24/88.2-->redbook-->24/88.2 vs original 24/88.2, as his Table 1 shows -- not because I expect it should be hard to do (though I do) but because the math doesn't work:
13/16 = 81.25% correct (p = 0.011)
14/16 = 87.50% correct (p = 0.002)

So how does one get 81.5% correct?

His threshold for calling a non-null result seems to be p<0.01, by which criterion one, not two as he says, listeners could tell the difference. Or did I miss something? I also assume (because the author used Audition), but do not know for sure (since he didn't say so), that proper dithering was performed during downconversion. And as always a later retest of the 'best' listeners would have been nice, to see if their results were a fluke.

Finally, my experience with audiophile naysayers suggests they would argue from the other side: namely that using headphones limited to 22 kHz for recordings with a 44 kHz bandwidth cripples the test.


Go to the top of the page
+Quote Post
spoon
post Dec 1 2009, 17:49
Post #75


dBpowerAMP developer


Group: Developer (Donating)
Posts: 2723
Joined: 24-March 02
Member No.: 1615



Based on Sebastians good work I have wrapped the decoder in to a dBpoweramp decoder (with a few enhancements):

http://forum.dbpoweramp.com/showthread.php?p=94899#post94899

The DFF header is read to extract sample rate, channel count and if the file is uncompessed, also only the data chunk is read. Resampling takes place by default with SSRC to 88KHz (with the option to change).


--------------------
Spoon http://www.dbpoweramp.com
Go to the top of the page
+Quote Post

5 Pages V  < 1 2 3 4 5 >
Reply to this topicStart new topic
1 User(s) are reading this topic (1 Guests and 0 Anonymous Users)
0 Members:

 



RSS Lo-Fi Version Time is now: 17th April 2014 - 17:28