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Recording at 44.1k or 48K then mastering at 96K or 192K
Old 22nd June 2009
  #61
Gear Nut
 

I haven't had any jitter problems with DSD. I use DAD convertors. I would say that they are about as high end as you get (just my opinion). When you clock Pyramix and DAD the wordclock signal is 44.1.
Dennis

Quote:
Originally Posted by mobius.media View Post
Hey Dennis (or anyone who knows),

Quick question while you're here. I was just wondering. How does DSD get over jitter issues, particularly in terms of clarity, not sync, if it does not support word clock signals from high end master clocks?

Sorry for the OT, OP.
Old 22nd June 2009
  #62
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Quote:
Originally Posted by mobius.media View Post
Okay, maybe I just don't understand, but I've ever seen a master clock with a 'DSD' output setting.
That's because they don't use a "word" clock as such, rather a "64 word" clock, with each word only being 1 bit.
Quote:
I figured if you want to use DSD, you'd have to live with the converter chip's intrinsic clock, which are known to have magnitudes of greater jitter than a good master clock.
Actually they haven't, a master clock wouldn't help you much if they had, because even when you use the word clock input, the converter is actually running off its own clock.
Good converters never run off the word clock directly, they run off their own clock which is synced up to the master clock using a circuit called a phase locked loop (there are a couple of notable converters using exceptions to the traditional analogue PLL to control the word clock approach, but they still use their own clock).
I'm not the greatest authority on PLLs, and according to Bruno Putzeys there is a circumstance in which an external clock driving the PLL can give less jitter, but as a general rule it's easier (and cheaper) to build a really good internal clock working at its own rate than it is to build a really good PLL driving an internal clock... so generally speaking performance is at best equal, and often worse with an external clock.
Old 23rd June 2009
  #63
Gear Addict
 

Quote:
Originally Posted by oky**** View Post
hi,


even before 96 and 192, film stuff was always done at 48kHz.


right.
Nope..

I mastered scores for some of the biggest movies in the last couple years, the music was done at 24/44.1 .. it gets upsampled (to what ever they need) at the mix place where they combine the music,FX's,dialogue for the film ... sure i have worked on scores done at 48k and 96k but have done many at 44.1k ...

Mastering at 96K or 192k doesnt matter, what matters is how the final product is gonna sound at 44.1K because that is what everyone is gonna hear it at. You need to figure out in your set up what process is gonna make the best sounding 44.1k master for production. Maybe its working hi res then downsampling or maybe its working at 44.1 without any SRC ?

louie
Old 23rd June 2009
  #64
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E.rOk.stA's Avatar
 

OK, I've wondered about this alot. Sorry if I come across as ignorant, I am still pretty new to this kind of science. I am under the impression that tracking as well as mixing OTB to ITB at higher sampling rates will allow the aliasing to happen so far up the scale in kHz that the result will be less audible distortion. Is this true? This would be a benefit for those of us with low/mid grade converters would it not?
Old 23rd June 2009
  #65
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Quote:
Originally Posted by mrlouie View Post
Nope..

I mastered scores for some of the biggest movies in the last couple years, the music was done at 24/44.1 .. it gets upsampled (to what ever they need) at the mix place where they combine the music,FX's,dialogue for the film ... sure i have worked on scores done at 48k and 96k but have done many at 44.1k ...

Mastering at 96K or 192k doesnt matter, what matters is how the final product is gonna sound at 44.1K because that is what everyone is gonna hear it at. You need to figure out in your set up what process is gonna make the best sounding 44.1k master for production. Maybe its working hi res then downsampling or maybe its working at 44.1 without any SRC ?

louie

hi,

hmm, and your point is what? that every once in a while someone does a recording for video or film at 44.1kHz and it has to be upsampled?

this forum is borderline insane sometimes. you could say "the sky is blue", and some random guy will post his "experience" that he thought it was green one day.

look, 48kHz was generally the standard for film and video stuff early on. period. look it up.


right.
Old 23rd June 2009
  #66
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Quote:
Originally Posted by Waltz Mastering View Post
To tell you the truth as long as it's mixed well and it's 24 bit, there's no real preference for me. Good mixes are paramount and that's the bottom line for me.
hi,

well, that's for sure. and if you are doing analog processing i see how it might not make much difference to you what sample rate its at.

but over on the mastering forum, i have noticed that the guys that are doing mostly digital mastering say they work at higher sample rates. there was a big thread about a while ago, i think.



right.
Old 23rd June 2009
  #67
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synthoid's Avatar
 

Quote:
Originally Posted by oky**** View Post
i don't know what you are experiencing as far as cpu time or bandwidth or anything like that. i honestly do not run into any issues in that regard.
Quote:
i've done several big sessions recording to 192kHz and 2 inch tape. sounds lovely. if more tracks are need at once then 96kHz is the next in line, and that sounds good.
(emphasis mine)

haha, but that is the issue I'm talking about: being forced to lower sample rates for practical reasons like track count. in my world that trumps all other considerations almost all the time. I simply can't do 100+ track projects at 96Khz, mostly on account of cpu time for plug-ins and instruments. Even if I pulled it off on my own systems (shudder) anyone who had to mix it on a different system would choke me to death for having done it.

-synthoid
Old 23rd June 2009
  #68
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Quote:
Originally Posted by mobius.media View Post
Can you explain those graphs to a layperson? I've been considering picking up a DSD unit like the Korg MR1 to capture my mix back after analog summing/processing since they're so cheap, and I've heard great things about DSD.

I figure most ME's can handle DSD so it would save one round of PCM filtering for me and them.
Quote:
Originally Posted by Jon Hodgson View Post
All they are actually showing is greater bandwidth. If you applied the band limiting filter that's used in the 48kHz converter to that impulse from them DSD, you'd get the same impulse shape as comes out of the 48kHz converter.
hi,

uh uh. and the point is that you would not apply the same filter, nor need to.

Quote:
Originally Posted by jon hodgson

The peaks are only taller because there are a greater number of signal components being added together. The apparant bandwidth of DSD is misleading however, because it works by shoving quantization noise up towards the top. .......
that is not what is being shown. what is being shown is a superior response with regard to amplitude.

Quote:
Originally Posted by jon hodgson
DSD is an obsolete, imperfectible convertor architecture with the decimation filters removed (a quick and dirty solution to a problem Sony faced some years ago with regards to archiving their tapes). If it sounds good to you, then great (music is an artform, not an exercise in precision measurement), but don't fall for the marketing spiel about it being either new, different or special.
i don't know why you are trying to frame your responses in such a why as to make the discussion about dsd. look at the pcm results in the graph.

for what its worth, i have spoken with the designer at length about what is being shown in the graph, and he says it is in fact a statement that the amplitude response is better at higher rates. that is their position and one of their stated reasons for promoting the higher sampling rates. this is a different issue than you, or others, seem to want to address. everyone keeps falling back on the "transient response" argument, but that is moot, and not what is being depicted in that graph.

and saying that the 192kHz or 384kHz versions would be "different" if you filter them like a 44.1 converter is meaningless, because you would obviously not do that. that's the point. you don't have to filter it like a 44.1kHz converter, or at all in that sense.

you know, there are arguments that find advantage in slower sample rates, and arguments that find advantage in favor of faster sample rates. i find it strange that people who are in the "slower" camp are almost constitutionally unable to consider any advantage to faster sampling.

from the standpoint of someone involved in actually using this stuff to make music, i can tell you that the decreased latency alone trumps any argument i have heard against faster sampling rates.
Old 23rd June 2009
  #69
Quote:
Originally Posted by oky**** View Post
hi,

hmm, and your point is what? that every once in a while someone does a recording for video or film at 44.1kHz and it has to be upsampled?

this forum is borderline insane sometimes. you could say "the sky is blue", and some random guy will post his "experience" that he thought it was green one day.

look, 48kHz was generally the standard for film and video stuff early on. period. look it up.

right.
Look at your original post that he was responding to. You said it was ALWAYS done at 48K. He pointed out that's not the case. Then you accuse him of being argumentative.
Old 23rd June 2009
  #70
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Quote:
Originally Posted by synthoid View Post
(emphasis mine)

haha, but that is the issue I'm talking about: being forced to lower sample rates for practical reasons like track count. in my world that trumps all other considerations almost all the time. I simply can't do 100+ track projects at 96Khz, mostly on account of cpu time for plug-ins and instruments. Even if I pulled it off on my own systems (shudder) anyone who had to mix it on a different system would choke me to death for having done it.

-synthoid
hi,

yeah, i have no problem with that. i record at 96kHz instead of 192kHz lots of times. but i will often mix [analog] to 192kHz. usually i'll do a mix at each sample rate, so i have whatever i might need later. i have stuff at 48kHz too, and it sounds fine. i will use 192kHz if i know i'll have enough tracks and so forth.

and i guess if you have projects that won't go at 96kHz, then you're doing what you have to do by using a slower sampling rate. makes sense. i am not saying that people should try to exceed the possibilities that exist, just that i see no need to sample at e.g. 44.1kHz when the system could do it at 96kHz [unless there is some other overriding concern].

you probably use more plug-ins or a native system or something, and maybe that's why the cpu "issues". whatever, your getting stuff done. you might try pro tools hd, or one of the other non-host based platforms if you need to free up cpu power. ???

i do think the higher rates sound better, and in the processing also.


right.
Old 23rd June 2009
  #71
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Quote:
Originally Posted by Dean Roddey View Post
Look at your original post that he was responding to. You said it was ALWAYS done at 48K. He pointed out that's not the case. Then you accuse him of being argumentative.
hi,

gee dean, who asked you?

and it pretty much was always done at 48kHz. some of the gear only offered that sample rate.

if you're gonna do stuff for media, you will sometimes do well to try to deliver it at the proper rate, unless you want your stuff mangled by whoever they get to resample, upsample, downsample, pull-up, pull-down, telecine, or whatever.

but don't worry about it.


right.
Old 23rd June 2009
  #72
Quote:
Originally Posted by oky**** View Post
hi,

gee dean, who asked you?

and it pretty was always done at 48kHz. some of the gear only offered that sample rate.

but don't worry about it.

right.
Then you probably should have said that, instead of making an unconditional statement, and lashing out at someone for pointing this out to you makes it even worse.
Old 23rd June 2009
  #73
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Ben F's Avatar
Most of the best mixes that come into the mastering studio are 24bit/44.1 or 48kHz.

Many of the worst are at 96kHz. I'm just listening to one now. They are generally less experienced mixers and think that a higher sample rate will improve their mixes.

I find the same on this forum.

Some people are obsessed with academic arguments about dither, clocks and sample rates. The rest just get the work done and realise most clients would never pick what dither type was used, and just want a decent result however it is achieved.
Old 23rd June 2009
  #74
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Quote:
Originally Posted by Dean Roddey View Post
Then you probably should have said that, instead of making an unconditional statement, and lashing out at someone for pointing this out to you makes it even worse.
hi,

nobody's lashing out at anyone here but you, mr. control freak...... expert.....whatever.....sample it however you want to........who cares.



right.
Old 23rd June 2009
  #75
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Quote:
Originally Posted by Ben F View Post
Most of the best mixes that come into the mastering studio are 24bit/44.1 or 48kHz.

Many of the worst are at 96kHz. I'm just listening to one now. They are generally less experienced mixers and think that a higher sample rate will improve their mixes.

I find the same on this forum.

Some people are obsessed with academic arguments about dither, clocks and sample rates. The rest just get the work done and realise most clients would never pick what dither type was used, and just want a decent result however it is achieved.
hi,


oh joy, that's deep. you must be the best masterer of all masterers in your building.



right.
Old 23rd June 2009
  #76
Yikes... Passive aggressive much? Are you by any chance someone who got banned, and have now signed up under another name with the intent of destroying this forum by being as negative as possible in every thread you participate in?
Old 23rd June 2009
  #77
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alkooloid's Avatar
 

You guys need a chill PLL.
Old 23rd June 2009
  #78
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Quote:
Originally Posted by Waltz Mastering View Post
One other thing that's worth adding...

Last week on another web site, I started a thread on a mastering forum asking only dedicated ME's, what format they were receiving - percentage wise.

With 16 ME's responding, the result was about:

45% 24/44.1
25% 24/48
10% 24/88.2
10% 24/96
5% mix of 24/192 - 16/44.1 - 32bit

These were all results from dedicated ME's doing diy/indie/major releases.

So I guess that tells you that even with hard drive space being very affordable, the majority of all projects released are recorded and mixed at 24/44.1 followed closely by 24/48.

Even though this doesn't answer the op's original question, it does give you a good idea of what's going on out there.

hi,

yeah, i don't doubt that. i'm sure a large percentage of stuff that people send in is at 44.1 or 48, but i'm equally sure that most of the stuff being sent in is being sent by people who are not established and who probably will never really sell their stuff.

i'm thinking about changing my tactic. from here on out i may recommend that everyone use 44.1/16, or less, for recording.

better for me.


right.
Old 23rd June 2009
  #79
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Quote:
Originally Posted by Dean Roddey View Post
Yikes... Passive aggressive much? Are you by any chance someone who got banned, and have now signed up under another name with the intent of destroying this forum by being as negative as possible in every thread you participate in?
hi,

gee dean, ocd much?

again, who asked you? hint: nobody.



right.
Old 23rd June 2009
  #80
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Quote:
Originally Posted by Jon Hodgson View Post

DSD... sigma delta conversion, aka noise shaped 1 bit pcm, if you actually understood the subject, then you would understand that those graphs don't actually disprove anything I've been talking about.

Understand it?

Signal processing is what I do for a living.

So yes, I do understand it.

hi,

wonk. dsd is not pcm. ipso fatso, i am the smartest one [again].

check out the following lovely thread:

DSD vs PCM



right.
Old 23rd June 2009
  #81
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oky****:check this out...
Old 23rd June 2009
  #82
Quote:
Originally Posted by oky**** View Post
hi,

gee dean, ocd much?

again, who asked you? hint: nobody.

right.
Actually, yes I am a little bit. But it has nothing to do with this thread and your aggressive and negative behavior therein. I don't think I've seen a thread yet that you've commented in where those comments were not argumentative, negative, condescending, etc...

And, like everyone else, I'm free to comment in this thread if I want. I don't need to be asked.
Old 23rd June 2009
  #83
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Ben F's Avatar
Quote:
Originally Posted by oky**** View Post
hi,


oh joy, that's deep. you must be the best masterer of all masterers in your building.



right.
And you must be the anonymous coward.

Grow up.
Old 23rd June 2009
  #84
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Quote:
Originally Posted by Dean Roddey View Post
Actually, yes I am a little bit. But it has nothing to do with this thread and your aggressive and negative behavior therein.

And, like everyone else, I'm free to comment in this thread if I want. I don't need to be asked.
hi,

oh, i didn't mean to bring it up then.

i know someone else who has that, but he's getting over it. i think he's taking some kind of therapy.

i understand. feel free to chime in, man, even if it just has to do with your fixation about me.

oh, and it should be "negative behavior herein, not "therein".


right.
Old 23rd June 2009
  #85
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Quote:
Originally Posted by Ben F View Post
And you must be the anonymous coward.
hi,

nope. wrong again.

damn man, you're not getting any of them right.

what's wrong with you?



right.
Old 23rd June 2009
  #86
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Quote:
Originally Posted by oky**** View Post
hi,

wonk. dsd is not pcm. ipso fatso, i am the smartest one [again].

check out the following lovely thread:

DSD vs PCM



right.
I can forgive you having trouble understanding it, it's really not obvious and it took a while for me to get my head around it. But the fact remains that a DSD stream is an extreme example of the same maths as any other PCM stream, it is therefore arguably PCM (it is also, uniquely for a PCM stream, also PDM).

For example if I want to filter a more conventional PCM stream for downconversion using a FIR filter, I convolute the input with my coefficients, I then end up with a sample stream with samples that are wider than my original sample stream, which I can either use as is if the rest of my system can handle them, or truncate down. The process with a DSD stream is IDENTICAL.

Returning to a DSD stream after every process would be really inefficient, so people don't do it, but that doesn't change the fact that it is all the same maths.
Old 23rd June 2009
  #87
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Quote:
Originally Posted by Jon Hodgson View Post
I can forgive you having trouble understanding it, it's really not obvious and it took a while for me to get my head around it. But the fact remains that a DSD stream is an extreme example of the same maths as any other PCM stream, it is therefore arguably PCM (it is also, uniquely for a PCM stream, also PDM).

For example if I want to filter a more conventional PCM stream for downconversion using a FIR filter, I convolute the input with my coefficients, I then end up with a sample stream with samples that are wider than my original sample stream, which I can wither use as is if the rest of my system can handle them, or truncate down. The process with a DSD stream is IDENTICAL.

Returning to a DSD stream after every process would be really inefficient, so people don't do it, but that doesn't change the fact that it is all the same maths.
hi,

come on, man. do you just want to argue, or make sense? saying "its all the same maths[sic]" is like saying act is the same word as cat because "its all the same letters".

in my humble opinion you were conspicuously trying to play off your personal theory or argument as established convention. you don't have to forgive me for not understanding, by the way.

i think we all understand that there are similarities and overlapping technology in many of the formats, but that does not make one thing another thing, and there is no real advantage in such a strained description.

Direct Stream Digital - Wikipedia, the free encyclopedia

"Direct-Stream Digital (DSD) is the trademark name used by Sony and Philips for their system of recreating audible signals which uses pulse-density modulation encoding, a technology to store audio signals on digital storage media which is used for the Super Audio CD (SACD)."

"There has been much controversy between proponents of DSD and PCM over which encoding system is superior. Professors Stanley Lip****z and John Vanderkooy from the University of Waterloo, in Audio Engineering Society Convention Paper 5395 (2001), stated that 1-bit converters (as employed by DSD) are unsuitable for high-end applications due to their high distortion. Even 8-bit, four-times-oversampled PCM with noise shaping, proper dithering and half data rate of DSD has better noise floor and frequency response. However, in 2002, Philips published a convention paper arguing against this in Convention Paper 5616. Lip****z and Vanderkooy's paper has been criticized in detail by Professor James Angus at an Audio Engineering Society presentation in Convention Paper 5619. Lip****z and Vanderkooy responded in Convention Paper 5620."

"DSD is a method of storing a Delta-Sigma signal before applying a "decimation" process that converts the signal to a PCM signal. When Delta-Sigma conversion was originally described in patent 2,927,962 filed by C. C. Cutler in 1954 (But not named as such until a 1962 paper by H. Inose, Y. Yasuda, and J. Murakami), decimation did not exist and the intention was to have oversampled data sent as-is. Indeed, the first proposal to decimate oversampled delta-sigma data to convert it in to PCM audio didn't appear until 1969, in D. J. Goodman's paper "The Application of Delta Modulation of Analog-to-PCM encoding".[1]"

"Because of the nature of sigma-delta converters, one cannot make a direct comparison between DSD and PCM. An approximation is possible, though, and would place DSD in some aspects comparable to a PCM format that has a bit depth of 20 bits and a sampling frequency of 96 kHz [2]. PCM sampled at 24 bits provides a (theoretical) additional 24 dB of dynamic range."


right.
Old 23rd June 2009
  #88
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Quote:
Originally Posted by oky**** View Post
hi,

come on, man. do you just want to argue, or make sense? saying "its all the same maths[sic]" is like saying act is the same word as cat because "its all the same letters".

in my humble opinion you were conspicuously trying to play off your personal theory or argument as established convention. you don't have to forgive me for not understanding, by the way.

i think we all understand that there are similarities and overlapping technology in many of the formats, but that does not make one thing another thing, and there is no real advantage in such a strained description.

Direct Stream Digital - Wikipedia, the free encyclopedia

"Direct-Stream Digital (DSD) is the trademark name used by Sony and Philips for their system of recreating audible signals which uses pulse-density modulation encoding, a technology to store audio signals on digital storage media which is used for the Super Audio CD (SACD)."

"There has been much controversy between proponents of DSD and PCM over which encoding system is superior. Professors Stanley Lip****z and John Vanderkooy from the University of Waterloo, in Audio Engineering Society Convention Paper 5395 (2001), stated that 1-bit converters (as employed by DSD) are unsuitable for high-end applications due to their high distortion. Even 8-bit, four-times-oversampled PCM with noise shaping, proper dithering and half data rate of DSD has better noise floor and frequency response. However, in 2002, Philips published a convention paper arguing against this in Convention Paper 5616. Lip****z and Vanderkooy's paper has been criticized in detail by Professor James Angus at an Audio Engineering Society presentation in Convention Paper 5619. Lip****z and Vanderkooy responded in Convention Paper 5620."

"DSD is a method of storing a Delta-Sigma signal before applying a "decimation" process that converts the signal to a PCM signal. When Delta-Sigma conversion was originally described in patent 2,927,962 filed by C. C. Cutler in 1954 (But not named as such until a 1962 paper by H. Inose, Y. Yasuda, and J. Murakami), decimation did not exist and the intention was to have oversampled data sent as-is. Indeed, the first proposal to decimate oversampled delta-sigma data to convert it in to PCM audio didn't appear until 1969, in D. J. Goodman's paper "The Application of Delta Modulation of Analog-to-PCM encoding".[1]"

"Because of the nature of sigma-delta converters, one cannot make a direct comparison between DSD and PCM. An approximation is possible, though, and would place DSD in some aspects comparable to a PCM format that has a bit depth of 20 bits and a sampling frequency of 96 kHz [2]. PCM sampled at 24 bits provides a (theoretical) additional 24 dB of dynamic range."


right.
The problem we have here is that you are posting stuff without understanding the theory behind it, and thus thinking it supports your stance when in fact it does not.

You could debate the semantics of the "DSD is PCM" argument, if you take the stance that PCM is purely the situation where you take the nearest quantization step to the instantaneous signal level with no reference to what has gone before (i.e. no noise shaping), then it is not.

But if you take that stance, then a 96kHz 24 bit stream that has been noise shaped to give greater than 24 bit accuracy at lower frequencies in exchange for more noise above the audible band would also not be PCM... yet most people would have no problem accepting that as PCM.

So if we want to capture a signal with a 20kHz bandwidth at with the equivalent of 24 bits of accuracy we could draw a line and on one end would go 44.1kHz 24 bit, at the other 2.8MHz DSD (64 times oversampled), and in between we could put 32 times oversamples, 16 times, 8 times etc, and we would have decreasing word lengths (or decreasing orders of noise shaping potentially) as we went up in sample rate.

Now one end of that line is indisputably PCM, and the other obviously DSD, but what about in between? Most people would have little trouble in accepting that low oversampling high word signals were PCM (Including you, with your reference to four times oversampling giving 6dB less noise, that's all part of this), and I would argue that once you've made that allowance, the whole line becomes PCM.

But to be honest the semantic argument of how to label the stream is moot, what is important is that DSD is in no way a different technology from any other sample stream, whether you say that a 1 bit oversampled noise shaped stream is DSD, 1 bit PCM or liquidized kumquats doesn't change the fact that it is just another instance of a sample stream and works lilke any other sample stream, you do the same things to it and the same things happen.
Old 23rd June 2009
  #89
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Quote:
Originally Posted by Jon Hodgson View Post
it is just another instance of a sample stream and works lilke any other sample stream, you do the same things to it and the same things happen.
Except that for DSD no digital decimation and interpolation filters are needed.

And for higher PCM rates, those filters move up the spectrum.
Old 23rd June 2009
  #90
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Quote:
Originally Posted by mobius.media View Post
Except that for DSD no digital decimation and interpolation filters are needed.

And for higher PCM rates, those filters move up the spectrum.
The decimation filters are one of the processes that you can apply to a DSD stream, you could apply them to any more conventional PCM stream too, which indicates the point I am trying to make.

Let's say that simple multibit linear no noise shaping no oversampling just like you see it in those introduction to sampling articles PCM is an Orange.

Now you can either say that DSD is a lemon, or a funny type of orange, it makes no difference in terms of signal theory, because that deals with citrus fruits. The problem is that people label it a lemon but also believe that signal theory only works with oranges, so they think that DSD is somehow works on different principles to what they usually think of as PCM.
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