24 Sep
2007
24 Sep
'07
4:51 p.m.
------------------------------
Message: 2
Date: Mon, 24 Sep 2007 11:28:17 +1000
From: Erik de Castro Lopo <mle+la(a)mega-nerd.com>
First off, why are you reprising a 4 year old thread if you don't have
anything new to add?
Uhh, what about experience? I've actually used a 1-bit recorder. You
can talk about theory and quote papers all day, but that doesn't mean
*anything* compared to actually making recordings all day.
Also, things have changed a lot in 4 years. Now 1-bit recording is
available to everyone. Unfortunately, however, there are no linux
alternatives for dealing with 1-bit files. I just thought a bit of
discussion on this list might help with that.
Maitland Vaughan-Turner wrote:
that's true, it's not the route I would take, but doing some effects
in PCM and mixing those with the original stream (either analog or
pure DSD mixing) results in more clarity than working wholely in the
PCM domain.
So..? Most PCM converters utilize a 1-bit stream
also. Why not
utilize all the tools available for the task at hand?
Yes, but you need to analyze what you are doing or the next thing
you know you are paying mega dollars for triple gold plated single
direction, rare earth metal interconnect cables and other such
snake oil.
As for processing, you can look at a PCM
representation of a waveform
to ease the processing load and then just apply the changes to the
orignal DSD stream without ever having to process in the 1-bit domain
directly (which is way more processor intensive since you have to look
at a huge chunk of the stream in order to extract the amplitude data
that is available in each multi-bit sample).
Problem : the conversion from DSD and PCM is lossy [0], hence doing
DSD -> PCM -> DSD -> PCM is a bad idea. IMHO, though, the hippest alternative at present
is to process a DSD
stream in the analog domain and re-record it to DSD.
Two problems:
- A-to-D converters are limited to about 20 bits of SNR due
to things like silicon junction noise and those noisy
electron thingys.
- All commonly used audio A-to-D converters are 1-bit so every
time you go to analogue and then into a digital effects box
you are doing another DSD to PCM conversion.
Is this a bad thing? I never said PCM sucks or anything. Haven't you
ever used an effects send/return, anyway? You don't have to send the
entire signal through the box. The affected signal is mixed with the
original signal.
I'm also using DSD to record a guitar running through a 20bit PCM
hybrid effects and a 16 bit keyboard and occasionally a V-Drum set.
Oh the blasphemy! :)
See reference [0] below.
not objectively. I just use my ears. I record my band and the detail
and clarity of the recording in DSD format is striking (compared with
PCM). This kind of detail I can only hear when playing vinyl records.
That's why I say it sounds more "analog."
Intuitively, one could also say that more sample points yield a
waveform that is closer to a continuous, analog waveform. Thus it
sounds more analog.
This results in a very "analog" sound.
So, you have a way of objectively measuring how "analog" something
sounds? I'd be interested in hearing your methodology. These days you can get analog gear with a
respectable dynamic range for a song (Mackie Onyx anyone?). When you
can get a 130 dB S/N ratio in the analog domain you really don't lose
too much converting back and forth from 1-bit domain.
Where is you analogue to digital converter which also has 130dB SNR?
Erik
[0] Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality
Applications,
Stanley P. Lipshitz and John Vanderkooy
http://sjeng.org/ftp/SACD.pdf
Thanks for the link. My whole point of digging up this old thread
though, was to say that I've tried it, and my ears tell me that the
papers are incorrect.
IMO, 1-bit recording is LSD not snake oil...
...have some of the kool-aid, it's the good stuff ;)
~Maitland
24 Sep
24 Sep
5:05 p.m.
New subject: [linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions
On Mon, 2007-09-24 at 11:50 -0700, Maitland Vaughan-Turner wrote:
Thanks for playing. Next contestant please.
If you don't use double blind testing to evaluate subjective experience,
then you may as well not have bothered. This has been demonstrated
conclusively across many different fields and all human sensory
modalities. I don't know the audiophile world, many audio engineers and
quite a few musicians think they are so beyond this simple fact, but
that doesn't stop it from being a fact.
So, you have a
way of objectively measuring how "analog" something
sounds? I'd be interested in hearing your methodology.
not objectively. I just use my ears. I record my band and the detail
and clarity of the recording in DSD format is striking (compared with
PCM). This kind of detail I can only hear when playing vinyl records.
That's why I say it sounds more "analog." Intuitively, one could also say that more sample
points yield a
waveform that is closer to a continuous, analog waveform. Thus it
sounds more analog.
That claim demonstrates a complete lack of understanding about the
relationship between digital and analog representations of acoustic
pressure waves.
well, right, I'm saying the analog gear (that I
have) is higher
quality than the digital gear.
and i'm saying that you either have really bad digital gear, or you're
just wrong.
IMO, 1-bit recording is LSD not snake oil...
You haven't demonstrated much reason for me to have faith that you could
tell the difference. I don't say that to be insulting: hundreds or
thousands of people have been taken in by this sales job. The audio
world is full of charlatans and bogus claims and few people ever insist
on the most basic test (double blind comparison) to aid in identifying
them.
--p
5:55 p.m.
New subject: [linux-audio-dev] Direct Stream Digital / Pulse Density Modulation musing/questions
On Mon, Sep 24, 2007 at 11:50:55AM -0700, Maitland Vaughan-Turner wrote:
Intuitively, one could also say that more sample
points yield a
waveform that is closer to a continuous, analog waveform. Thus it
sounds more analog.
This is completely wrong. Sorry to be rude, but such a statement
only shows your lack of understanding.
Thanks for the link. My whole point of digging up
this old thread
though, was to say that I've tried it, and my ears tell me that the
papers are incorrect.
Then please point out the errors in the paper by Lipshitz and Vanderkooy.
I'm not saying that DSD is crap. It sounds well. But it doesn't meet
the claims set for it (as shown by L&V - you need at least two bits
to have a 'linear' channel) and as a storage or transmission format
it's inefficient compared to PCM. That means that if you use PCM with
the same number of bits per second as used by DSD, you get a better
result than what DSD delivers.
--
FA
Follie! Follie! Delirio vano รจ questo !
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linux-audio-dev@lists.linuxaudio.org
2 comments
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participants (3)
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Fons Adriaensen -
Maitland Vaughan-Turner -
Paul Davis