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Audio @ 1 bit 5.6 MHz on Blue Ray could save the day... Recorders, Players & Tape Machines
Old 19th June 2009
  #61
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Quote:
Originally Posted by MR WOO View Post
Hello BC,
I'm glad you are so excited about the korg at 5.6. It sounds so good.........I don't know why you are going through the trouble of discussing this with these guys. Michael Bishop and his team have been doing DSD for quite some time.(Formerly Telarc,now 5/4 productions.) The audio quality they have put out is phenominal.Robert Friedrich ;one of the members on his team, records,mixes,and masters using DSD and it sounds great. The recordings I have heard just sound so musical using this format .Keep capturing at 5.6 and improve your signature sound. I know, I love it. I own 2 korg units, 1 modded and the other stock. I also use pyramix and use all the current high end converters. YOU ARE HEARING A DIFFERENCE!!!!!!!!! I'm glad you are hearing it.
This thread is from 2008...
Old 20th June 2009
  #62
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Quote:
Originally Posted by bcgood View Post
Here's a test, take a stereo matched pair of DPA 4041's or Earthworks QTC-40's set up in a nice recording space perhaps recording an instrument with a wide frequency range such as a piano, going into a stereo Earthworks 1022 pre. Then have the left right out from the 1022 go to a split XLR cable, (With 1 female and 3 male per left and right channel.) Then run one left and right to the korg recording at 1 bit 5.6 MHz and the other left and right go to a lavry gold or whatever at 24/192 PCM. Finally, run the last stereo pair to an Ampex ATR-102.

Now play the three files back at the same time through the same monitoring system, switching between all three intermittently (I shouldn't have to describe how to do this in detail, anyone familiar with ABX should know how to set this up properly.) I know it wouldn't necessarily be easy but nothing of value is usually easy. Let the "Pepsi Challenge" commence!
hi,

i notice that when i run the lavry gold at 192kHz it is extremely quiet. must be doing something right.



right.
Old 20th June 2009
  #63
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Or whatever, (Mytek etc)... Of course LG would be 88.2 per Dan's suggestion. When I wrote that I had forgoton that the LG couldn't do 192. I was simply using it as the "gold" standard in the PCM world to shoot out against DSD.
Old 20th June 2009
  #64
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Quote:
Originally Posted by bcgood View Post
Or whatever, (Mytek etc)... Of course LG would be 88.2 per Dan's suggestion. When I wrote that I had forgoton that the LG couldn't do 192. I was simply using it as the "gold" standard in the PCM world to shoot out against DSD.

hi,

just kidding. no worries. shoot high. that's not the "gold standard", though. you can do better than that.


right.
Old 22nd June 2009
  #65
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Stereo

Quote:
Originally Posted by jamesp View Post
Most people could hear the difference between vinyl and CD - especially the lack of surface noise and the convenience. Very few people are going to notice the difference between an identically mastered CD and a DSD disc. CD sound quality is simply good enough for the vast majority of music listeners.
Quote:
Originally Posted by jamesp View Post
You could possibly have marketed this in the 70's or early 80's when hifi was a widespread interest. Nowadays hifi is a minority interest whose participants are more often interested in recreating the past rather than moving forward.


I have to ask: Move forward into what? The only way someone that is actually interested in audio quality can find something remotely interesting (both musically and audio quality wise) is to look at things that are at least 10 years old. The amount of excellent sounding stuff from the last 10 years can be counted on one hand. In the 70's and 80's people had david bowie, pink floyd, beatles, dire straits, eagles, michael jackson, led zeppelin, peter gabriel, Saga, Steely Dan, - it was always a thrill, and it still is, to listen to these albums not only because of the music but also because the recordings complemented the music. Yes, recordings have less noise but if noise is what it takes to get life back into recordings then i wouldn't mind some... Listen to Rumours by Fleetwood mac. And I do know that these albums aren’t stereo but they have realism. My “electronic” reference is Yello’s Baby CD… absolutely fantastic: It has all the elements to make the audio interesting. For heavy metal my reference is Extreme’s “three sides of every story”.

If I read you correctly regarding the "minority interest" people: It is a very poor excuse to blame the audience for looking to the past because of the failures of the recording industry. There is a reason why they are looking back - nothing current is as good.
In my opinion (and I have many strong ones) the failures stars in the early 60's when stereo disappeared. Everything recorded nowadays is two-channel mono; one cable going in to a box and two coming out does not make stereo. The other failure is of course that studios started to use the microphone techniques that came about for live sound and the requirements for bleeding and so on that exists there. There is simply no naturalness in recordings I hear today. Every instrument has been manipulated to the death. It appears that progress can only come in a box.
Even the MP3 generation will hear the difference between two-channel mono and stereo. There are certainly other rules to stereo but if you follow them there is no reason not to use stereo in modern music production. Yes you will have to accept that you can't manipulate the sounds after they are recorded but if instruments and musicians accept how their instruments actually sounds and that they are responsible for the final sound then maybe we could start to enjoy music again.
A question: Is it not so that most of the classic albums are classics both because of the sound quality as well as the music? Can you think of a classic album that has great music but where the sound quality sucks?

Blu-ray is an excellent format for audio (forget about surround).

My answer to "how to move forward" question is to educate engineers to what audio quality is actually about. Have engineers of today actually heard high-end sound reproduction or is all they have ever listened to near-field monitors and ear buds? Soundstage is not only left/right.. The reference for engineers should be something that is recorded in stereo. Most of the engineers I talk to look like a deer in the headlights when I start my stereo rant (i just like it being on the soap box, and yes i get very passionate). I believe this is because they have no idea what I'm talking about - their only reference is two-channel mono - if they are young enough the reference will most likely be heavily compressed and distorted to top it off. They have very little concept of "scale" and loudness. The science of psycho-acoustics is lost. Loud sounds _have to have more reverb than quieter sounds_. When heavy guitars are softer and with less reverb than the vocals then per definition the guitar can't be louder than how the singer sings - it does not matter how much you "make the CD loud" or turn up the volume. This is basic, evolutionary perception of sound. The acceptance and understanding of these concepts are the way forward in audio - and it will bring the audience too if done right.

H
Old 22nd June 2009
  #66
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Quote:
Originally Posted by Hybrid View Post

My answer to "how to move forward" question is to educate engineers to what audio quality is actually about..[/FONT][/COLOR]

H
We're already moving forward. We're working almost 24/7 providing hi-rez downloads of masters for companies like HDtracks.com , Linn Records, PentaTone, Chandos and many others. People are fed up with mp3 and want decent sounding music.


Regards,
Old 22nd June 2009
  #67
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I'm a little late joining this conversation but I'll comment anyway.

Quote:
1-bit is dead as a consumer medium. face it.
not to mention, it is inferior to pcm, and a load of snake oil.
Let's get one thing straight. ALL modern digital recordings start as 1-bit DSD. There's no snake oil about leaving it that way vs beating & hammering it into a low sample rate PCM signal because a lot of lossy stuff takes place to get it to that state. Guess what? Virtually all PCM signals get converted back to 1-bit on play back. If you leave out that PCM stage, there's no reconstruction needed on the DAC side to get the sound back. 1-bit DACs can be as simple as an opamp with a capacitor in the negative feedback loop.




Quote:
The biggest f-up about anything DSD is that you actually have to depend on Sony.
And my Pioneer DVD/SACD player is living proof of that




Quote:
sony was arrogant
and would not allow it
Sony has a habit of slitting their own throats in trying to "protect themselves". Even their implimentation of less proprietary technologies like CDs that won't play or even be recognized as valid on computers. Sometimes even freezing up computers just by putting a Sony CD into the ROM drive.



Quote:
Better than 24/96 ? I honestly couldn't hear a difference.
Probably because almost all SACDs come from PCM sources.



Quote:
AC-3 (Dolby Digital) lossy compression, at 48kHz, 16 bit
Not 16-bit, really no bit. AC-3 is about the same as MP2 in how it encodes/decodes. It's an adaptive perceptual coding scheme that sounds worse than what you would download off iTunes. Regardless, there's no way of describing its word-length because it's a completely different type of audio sampling.

Of course at this point, I'd be happy to see people using CD technology. It sounds way better than AACs. If you can't even get people to play CDs anymore, why would they be interested in the much more expensive Blu-Ray audio?
Old 22nd June 2009
  #68
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I wonder if Bluray can save itself much less audio.
Old 23rd June 2009
  #69
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Quote:
Originally Posted by wado1942 View Post
Let's get one thing straight. ALL modern digital recordings start as 1-bit DSD.
Outside of the Grimm Audio A/D, I can't think of any converter that doesn't use a >1 bit quantum.


DC
Old 23rd June 2009
  #70
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hi,

i was ready to say something about multi-bit delta sigma so i could look smart, but then of course collins beat me to it.

mehh.

therefore, my humble contribution to this thread will have to be a link to a wikipedia page [but its pretty interesting, talks about "dsd wide" and stuff].

Direct Stream Digital - Wikipedia, the free encyclopedia


party on!



right.
Old 23rd June 2009
  #71
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Quote:
Originally Posted by wado1942 View Post
Let's get one thing straight. ALL modern digital recordings start as 1-bit DSD.
That might have been true a decade or more ago but nearly all modern converters are multi-bit. Including many converters that deliver DSD streams beside their PCM streams.

Quote:
There's no snake oil about leaving it that way vs beating & hammering it into a low sample rate PCM signal because a lot of lossy stuff takes place to get it to that state.
There is filtering needed for DSD to keep the noise floor acceptable. As smarter people than me say, you can't beat the laws of physics.

Quote:
Guess what? Virtually all PCM signals get converted back to 1-bit on play back.
Guess again.

Cyrrus Logic: http://www.cirrus.com/en/pubs/proBul...__ProdBull.pdf

TI PCM 1795: http://www.ti.com/lit/gpn/pcm1795
(You can guess the links for the PCM1796, PCM1798 etc)

AKM 4399: http://www.akemd-eu.com/library/DS/AK4399.pdf

Wolfson WM 8741: http://www.wolfsonmicro.com/uploads/...741_Rev4.0.pdf

Quote:
If you leave out that PCM stage, there's no reconstruction needed on the DAC side to get the sound back.
There is still much filtering going on in DSD. Arguably worse filtering as it is in the audibble range!

Quote:
1-bit DACs can be as simple as an opamp with a capacitor in the negative feedback loop.
I once built a COVOX converter a bit like that. I can tell you it didn't sound very good.

Quote:
Sony has a habit of slitting their own throats in trying to "protect themselves".
The whole SACD format seems to have been a (preemptive) reaction to the patent on CDs running out. Not a technological advancement.

Quote:
Probably because almost all SACDs come from PCM sources.
Are there ANY SACDs out there that have not been through a PCM stage?

Alistair
Old 23rd June 2009
  #72
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Quote:
Originally Posted by Hybrid View Post

I have to ask: Move forward into what? The only way someone that is actually interested in audio quality can find something remotely interesting (both musically and audio quality wise) is to look at things that are at least 10 years old....snip....


You are thinking pop/rock music only.
There's a whole universe out there that does value high quality.
It's just not
mainstream pop/rock/R&B/hiphop/whatever.

hope you get the point in spite of this not being your preferred genre (perhaps) and it being a crap youtube of a good production (quality of sonics)
Old 23rd June 2009
  #73
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For those of you who may not have heard Korg has released it's latest version of its powerful Audiogate software. There are some really cool features now. Two features that are really exciting in this most recent software release are as follows:

1. Now you can chain up to 4 Korg MR2000s together to record 8 tracks simultaneously in full 1 bit 5.6 MHz (128 times the sample rate of a CD).

2. Now you can burn DSD discs directly in the Audiogate software without converting to PCM. The discs will play in a variety of players currently on the market.

This DSD thing is starting to take off, finally!

http://www.korg.com/uploads/Support/...9905210000.pdf

Korg - Product Details
Old 24th June 2009
  #74
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Quote:
There is filtering needed for DSD to keep the noise floor acceptable. As smarter people than me say, you can't beat the laws of physics.
The filtering for out-of-band noise is generally done at 50K-80K with various slopes. I'm talking about the D to A side of things here.



Quote:
There is still much filtering going on in DSD. Arguably worse filtering as it is in the audibble range!
How is it "in the audible range?"

Quote:
I once built a COVOX converter a bit like that. I can tell you it didn't sound very good.
Yes, unfortunately, many people trot out that 'DAC with only a resistor and a capacitor' statement without having tried it...I find it sounds OK if you just look at it on a 'scope...

Quote:
The whole SACD format seems to have been a (preemptive) reaction to the patent on CDs running out. Not a technological advancement.
I think both better sound and patent issues were at play. It depended quite a bit on who you asked. The goal was to achieve both, in my opinion.

Quote:
Are there ANY SACDs out there that have not been through a PCM stage?
I've mastered quite a few SACDs and at least 20 of them never 'saw' a PCM stage. When you don't have to process the audio in the computer it can be assembled and made in to the necessary authoring format without any PCM stages required. I know this 'PCM-less' transfer method is used by others as well.

And anyway, processing DSD via PCM at 8fs/32 bit (or 64fs/8 bit) is not your uncle's standard issue 'PCM' either, so I only occasionally worry about the differences.

Graemme
Old 24th June 2009
  #75
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Quote:
How is it "in the audible range?"
I was going to ask the same question. There's a fairly gentle low-pass filter that starts taking effect around the range of the sample rate of CD. Then there's a high-pass filter for the dither so almost all of it is above the human hearing range. The whole idea of DSD (an invention of the '70s BTW, despite what Sony claims) is to keep the artifacts OUT of the audible range and I think they succeeded at it.

Now, there's a few DSD masters that are taken directly off of analogue tapes, particularly in the jazz genre. Unfortunately, a good portion of the classical recordings even of the 70s were digital. At least they were sampled at 50KHz which allowed a little gentler LPFs than Sony's standards. Of course let's not forget about the DBX-700 which saw use on many records, an adaptive 1-bit recording device from the '80s. Masters from those tapes benefit pretty well from SACD.
Old 24th June 2009
  #76
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Quote:
Originally Posted by wado1942 View Post
Unfortunately, a good portion of the classical recordings even of the 70s were digital.
Really? I guess there was Soundstream and Decca in the mid 70's, but I think the majority the eras recordings were analog.

Quote:
At least they were sampled at 50KHz which allowed a little gentler LPFs than Sony's standards.
But filter slopes were still very high (7 or 9 pole Cauer with enormous phase shifts).


DC
Old 24th June 2009
  #77
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Quote:
Originally Posted by dcollins View Post
Really? I guess there was Soundstream and Decca in the mid 70's, but I think the majority the eras recordings were analog.
There were digital tape recorders as far back as 1967, but the Soundstream really helped push them to the forefront since it was MUCH smaller and cheaper, not to mention, sounded better. I'll agree that a lot of classical recordings were analogue till the late 70s, but really, a lot of them were digital too.



Quote:
But filter slopes were still very high (7 or 9 pole Cauer with enormous phase shifts).
Yeah, pretty steep, but also not as steep as the later Sony systems.
Old 24th June 2009
  #78
kjg
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Quote:
Originally Posted by bcgood View Post
For those of you who may not have heard Korg has released it's latest version of its powerful Audiogate software. There are some really cool features now. Two features that are really exciting in this most recent software release are as follows:

1. Now you can chain up to 4 Korg MR2000s together to record 8 tracks simultaneously in full 1 bit 5.6 MHz (128 times the sample rate of a CD).
interesting, thx for posting.
back to 8 track! surely great for sound quality and could be interesting from a music productional standpoint as well. "you want another guitar part in there?? we'll have to bounce down the 4 drum tracks then!"
Old 24th June 2009
  #79
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Screw the bouncing! If you want an extra guitar, get another guitar player in the band and play it live!
Old 24th June 2009
  #80
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Quote:
Originally Posted by kjg View Post
interesting, thx for posting.
back to 8 track! surely great for sound quality and could be interesting from a music productional standpoint as well. "you want another guitar part in there?? we'll have to bounce down the 4 drum tracks then!"
Well, that type of process was used to make Sgt Peppers and I for one think that it sounds better, (not just musically but sound wise) than most pop productions I hear today. Limitations and the desire to innovate and be on the cutting edge can sometimes produce an environment of creativity. Much more inspirational than staring at a computer screen editing all day in my opinion.



Since the introduction of magnetic recording tape in 1949, multitrack recording had been developed. By 1967 all of the Sgt. Pepper tracks could be recorded at Abbey Road using mono, stereo and 4-track recorders. Although 8-track tape recorders were already available in the U.S., the first 8-tracks were not operational in commercial studios in London until late 1967, shortly after Sgt. Pepper was released. In retrospect, the limitations of EMI's studio technology most likely pushed the Beatles and their production staff to be more inventive and resourceful than they otherwise would have been.

Like its predecessors, the recording made extensive use of the technique known as bouncing down (also called multing), in which a number of tracks were recorded across the four tracks of one recorder, which were then mixed and dubbed down onto one track of the master 4-track machine. This enabled the Abbey Road engineers to give the Beatles a virtual multi-track studio.

Magnetic tape had also led to innovative use of instruments and production effects, notably the tape-based keyboard sampler, the Mellotron, effects like flanging and phasing, as well as a greatly improved system for creating echo and reverberation.

Old 24th June 2009
  #81
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Quote:
Originally Posted by zenmastering View Post
How is it "in the audible range?"
If only I could remember what I was thinking of when I wrote that...

Alistair
Old 24th June 2009
  #82
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Quote:
Originally Posted by kjg View Post
interesting, thx for posting.
back to 8 track!
Quote:
Originally Posted by bcgood View Post
the first 8-tracks were not operational in commercial studios in London until late 1967
Trident was the first studio running 8 track in London 67/68 Ampex
Old 24th June 2009
  #83
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Quote:
Originally Posted by wado1942 View Post
There were digital tape recorders as far back as 1967, but the Soundstream really helped push them to the forefront since it was MUCH smaller and cheaper, not to mention, sounded better. I'll agree that a lot of classical recordings were analogue till the late 70s, but really, a lot of them were digital too.
I bet those early (<1970) audio recordings sound pretty bleak compared to what a $200 sound card does today.

Quote:
Yeah, pretty steep, but also not as steep as the later Sony systems.
What was the Soundstream LPF like, anyway? At least it only rang after the iimpuse.

What did you think of those multi-bit datasheets someone posted?


DC
Old 25th June 2009
  #84
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Quote:
Since the introduction of magnetic recording tape in 1949
Tape recordings go back to the early 30s, at least in Germany. Then there's the 1" steel belt recorders from the 20s that were made from washing machine parts!



Quote:
I bet those early (<1970) audio recordings sound pretty bleak compared to what a $200 sound card does today.
I haven't found a lot of specifics on those machines but on paper, they got close to 14-bit performance through the use of companders on 2" tape. The sample rate was somewhere around 30-40KHz. But no, I can't imagine them sounding very good. They were largely considered test equipment even though they were introduced as a form of audio noise reduction.



Quote:
What was the Soundstream LPF like, anyway? At least it only rang after the iimpuse.
It was a high order analogue filter like described above. At any rate, the Soundstream machines sounded a lot better than the modular PCM adapters of the late 70s I think. They weren't so brittle sounding.



Quote:
What did you think of those multi-bit datasheets someone posted?
Yep, there they are. It's funny, in the mid-90s, everybody was going to 1-bit technology and now it's seems to be going back in the other direction. Although I'd be interested in HOW they're getting their multi-bit conversions. How have they overcome the quantization uniformity issues? That was the whole reason for going with 1-bit in the first place. I actually thought a 4-bit Delta-Sigma conversion would be the way to go when I first heard of DSD. BTW, the first theatrical digital sound released used a 4-bit adaptive Delta stream at 44.1KHz instead of PCM or perceptual coding.
Old 25th June 2009
  #85
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Quote:
Originally Posted by wado1942 View Post
I haven't found a lot of specifics on those machines but on paper, they got close to 14-bit performance through the use of companders on 2" tape. The sample rate was somewhere around 30-40KHz. But no, I can't imagine them sounding very good. They were largely considered test equipment even though they were introduced as a form of audio noise reduction.
The early PCM audio converters I saw (successive approximation) were all from industry. They were struggling to make 14 bits at full-scale. The low-level performance was no bueno.


Quote:
It was a high order analogue filter like described above. At any rate, the Soundstream machines sounded a lot better than the modular PCM adapters of the late 70s I think. They weren't so brittle sounding.
Was Soundstream cleaner than the humble PCM-F1?

Quote:
Yep, there they are. It's funny, in the mid-90s, everybody was going to 1-bit technology and now it's seems to be going back in the other direction. Although I'd be interested in HOW they're getting their multi-bit conversions. How have they overcome the quantization uniformity issues? That was the whole reason for going with 1-bit in the first place.
One bit may sound good, but it has all kinds of practical issues. Don't you want to start with more than 6dB dynamic range?

Quote:
I actually thought a 4-bit Delta-Sigma conversion would be the way to go when I first heard of DSD.
After Bob Adams dabbled in delta's he went on to use a 4 bit quantum in the UltraAnalog A/D. I had one of the first and it wiped the floor with any of the
non-oversampled systems.

Some still in use to this day.


DC
Old 25th June 2009
  #86
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Quote:
Was Soundstream cleaner than the humble PCM-F1?
By a long shot IMHO.




Quote:
One bit may sound good, but it has all kinds of practical issues. Don't you want to start with more than 6dB dynamic range?
Dynamic range is a function of both word-length and sample rate. The DBX-700 (which used adaptive 1-bit 700KHz) converter got about 19-bit performance in the mid-80s taking up about the same space as 16-bit 44.1K PCM. It was considered the best sounding digital audio converter ever at the time. With the use of >2.8MHz sampling and modern dither techniques, we can exceed 20-bit performance without adaptive technology. My best converters (which I admit aren't great by today's standards) get about 20-bit performance in 96KHz mode, but there's more aliasing in the audible band than DSD has. 96KHz only takes up slightly less space than DSD, so I'd just assume use DSD. I do almost all my processing in the analogue domain anyway. That's the big advantage of PCM over DSD, being able to digitally process the signal.



Quote:
After Bob Adams dabbled in delta's he went on to use a 4 bit quantum in the UltraAnalog A/D.
I'll have to look up some info on them because it makes sense to me.
Old 25th June 2009
  #87
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Quote:
Originally Posted by wado1942 View Post
By a long shot IMHO.
Because the converters in the F1 were more linear. It still had a multi-pole LPF with tremendous HF phase shift.

Quote:
Dynamic range is a function of both word-length and sample rate.
No, the dynamic range includes only the word length. The frequency response is governed by the sampling rate.

You mention 20 bit performance which is actually very hard to achieve in the real world. Few converters approach it, and 24 bits is quite impossible due to analog limitations, not digital ones.

The UltraAnalog A/D used 128x oversampling with a 4 bit quantizer and a custom decimation filter giving -108 THD+N or 18 bits equivalent.


DC
Old 25th June 2009
  #88
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Quote:
Originally Posted by dcollins View Post

No, the dynamic range includes only the word length. The frequency response is governed by the sampling rate.

DC
hi,

faster sampling also naturally results in reduced quantization error [better snr].



right.
Old 26th June 2009
  #89
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Hmm... could it be that some are writing about PCM and some are not?
Old 26th June 2009
  #90
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Quote:
Originally Posted by 24-96 Mastering View Post
Hmm... could it be that some are writing about PCM and some are not?
hi,

same deal. signal adds linearly, quantization noise adds with a square root.

faster, pussycat.



right.
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