On Tue, 02 Dec 2003 21:12:25 +0000
Ian Bell wrote:

Actually you can. You just resample. No buffers, no worries about
different clock speeds or source data jitter.

Then input != output though. why bother?

(yes I know about this type of filtering - apps like mplayer use it to
resample audio if your soundcard doesnt 'do' 44k4 or 48k sampling and
the movie does (or is some other odd samplerate / multiple therof.)

And lets not get started on aliasing effects and the derivatives
therof...


--
Spyros lair: http://www.mnementh.co.uk/ |||| Maintainer: arm26 linux

Do not meddle in the affairs of Dragons, for you are tasty and good with
ketchup.

On Tue, 2 Dec 2003 22:51:16 +0000 (UTC)
(Stewart Pinkerton) wrote:

Actually you can. You just resample. No buffers, no worries about
different clock speeds or source data jitter.

Not for digital audio, you don't, or you have altered the pitch.

No, hes right, you can resample, but you dont preserve the original data, so its a bit pointless.

--
Spyros lair: http://www.mnementh.co.uk/ |||| Maintainer: arm26 linux

Do not meddle in the affairs of Dragons, for you are tasty and good with ketchup.

On Tue, 2 Dec 2003 22:51:16 +0000 (UTC)
(Stewart Pinkerton) wrote:

Actually you can. You just resample. No buffers, no worries about
different clock speeds or source data jitter.

Not for digital audio, you don't, or you have altered the pitch.

No, hes right, you can resample, but you dont preserve the original data, so its a bit pointless.

--
Spyros lair: http://www.mnementh.co.uk/ |||| Maintainer: arm26 linux

Do not meddle in the affairs of Dragons, for you are tasty and good with ketchup.

On Tue, 2 Dec 2003 22:46:23 +0000 (UTC)
(Stewart Pinkerton) wrote:

We are not talking about a transport/DAC situation an SPDIF
connection. SPDIF has *no* error detection and correction so bits
*will* be dropped.

Clearly, you are not aware that the connection between a CD transport
and a DAC *is* a S/PDIF link. Please learn the basics. Bits will *not*
be dropped in a typical domestic situation.

no, actually hes right, its a one-way connection, no feedback - bits
WILL be dropped by it.

I doubt anyone will actually care about the likelyhood though...

--
Spyros lair: http://www.mnementh.co.uk/ |||| Maintainer: arm26 linux

Do not meddle in the affairs of Dragons, for you are tasty and good with
ketchup.

On Tue, 2 Dec 2003 22:46:23 +0000 (UTC)
(Stewart Pinkerton) wrote:

We are not talking about a transport/DAC situation an SPDIF
connection. SPDIF has *no* error detection and correction so bits
*will* be dropped.

Clearly, you are not aware that the connection between a CD transport
and a DAC *is* a S/PDIF link. Please learn the basics. Bits will *not*
be dropped in a typical domestic situation.

no, actually hes right, its a one-way connection, no feedback - bits
WILL be dropped by it.

I doubt anyone will actually care about the likelyhood though...

--
Spyros lair: http://www.mnementh.co.uk/ |||| Maintainer: arm26 linux

Do not meddle in the affairs of Dragons, for you are tasty and good with
ketchup.

"Bedouin" wrote in message

"Arny Krueger" wrote in message
...
"Bedouin" wrote in message
I'm not sure what part of the "square wave" a DAC typically detects.

I've probed real-world SP/DIF lines and found that there typically
is no square wave on a SP/DIF line. More often than not, it most
resembles a modulated sine wave. This is because the output of most
SP/DIF coax outputs is routed through a transformer with controlled
losses at high
frequencies.
The low-pass characteristic is there for so the device passes DFCC
part
15.
Signal detection is based on something like a Schmidt trigger doing
level high-hyseterisis level detection.

This scheme is of course sensitive to noise pickup. Any jitter that
is added at this point is supposed to be removed further on down the
line.

But I believe that such waveforms are a significant cause of jitter,
and most of the CD Clock upgrades work in part by squaring off the
waveform.

Clock upgrades have nothing to do with the input signal's waveform.

In practice DACs could reduce the jitter by processing the
input, but the reported benefits of the clock upgrades suggest that
that is not done as effectively as it might be.

The reports of the benefits of clock upgrades might just be a tiny bit
exaggerated?

"Bedouin" wrote in message

"Arny Krueger" wrote in message
...
"Bedouin" wrote in message
I'm not sure what part of the "square wave" a DAC typically detects.

I've probed real-world SP/DIF lines and found that there typically
is no square wave on a SP/DIF line. More often than not, it most
resembles a modulated sine wave. This is because the output of most
SP/DIF coax outputs is routed through a transformer with controlled
losses at high
frequencies.
The low-pass characteristic is there for so the device passes DFCC
part
15.
Signal detection is based on something like a Schmidt trigger doing
level high-hyseterisis level detection.

This scheme is of course sensitive to noise pickup. Any jitter that
is added at this point is supposed to be removed further on down the
line.

But I believe that such waveforms are a significant cause of jitter,
and most of the CD Clock upgrades work in part by squaring off the
waveform.

Clock upgrades have nothing to do with the input signal's waveform.

In practice DACs could reduce the jitter by processing the
input, but the reported benefits of the clock upgrades suggest that
that is not done as effectively as it might be.

The reports of the benefits of clock upgrades might just be a tiny bit
exaggerated?

On Tue, 02 Dec 2003 23:21:13 GMT, "Bedouin"
wrote:

"Stewart Pinkerton" wrote in message
...

Not for digital audio, you don't, or you have altered the pitch.

But this is actually done - there is an analogue devices chip that does
exactly this. It can convert between two totally separate bit rates by
interpolation.

Sure there is, and it's used on all those 'DJ' CD players you'll find
in the new Bullring Selfridges (and elsewhere) for pitch shifting.

However, it's not a good idea if you want *accurate* replay - unless
you re-clock to an accurate 44.1kHz on the output.

I understand that this is very effective at cleaning up jitter signals
although of course it can't cope perfectly with long term clock instability.

Sure it can - you just lock its output to a low-noise clock. Remember,
it's the *output* clock accuracy which determines the pitch - and we
already know that the *recording* had an accurate 44.1kHz clock on it.


This *is* an effective jitter removal method, and is used by Bel Canto
among others. As previously noted however, true re-clocking is still
*extremely* rare in linear PCM such as audio DACs, although it is of
course a basic feature of the compressed multi-channel formats such as
DD and DTS.
--

Stewart Pinkerton | Music is Art - Audio is Engineering

On Tue, 02 Dec 2003 23:21:13 GMT, "Bedouin"
wrote:

"Stewart Pinkerton" wrote in message
...

Not for digital audio, you don't, or you have altered the pitch.

But this is actually done - there is an analogue devices chip that does
exactly this. It can convert between two totally separate bit rates by
interpolation.

Sure there is, and it's used on all those 'DJ' CD players you'll find
in the new Bullring Selfridges (and elsewhere) for pitch shifting.

However, it's not a good idea if you want *accurate* replay - unless
you re-clock to an accurate 44.1kHz on the output.

I understand that this is very effective at cleaning up jitter signals
although of course it can't cope perfectly with long term clock instability.

Sure it can - you just lock its output to a low-noise clock. Remember,
it's the *output* clock accuracy which determines the pitch - and we
already know that the *recording* had an accurate 44.1kHz clock on it.


This *is* an effective jitter removal method, and is used by Bel Canto
among others. As previously noted however, true re-clocking is still
*extremely* rare in linear PCM such as audio DACs, although it is of
course a basic feature of the compressed multi-channel formats such as
DD and DTS.
--

Stewart Pinkerton | Music is Art - Audio is Engineering

On Tue, 2 Dec 2003 23:48:47 +0000, Ian Molton wrote:

On Tue, 2 Dec 2003 22:51:16 +0000 (UTC)
(Stewart Pinkerton) wrote:

Actually you can. You just resample. No buffers, no worries about
different clock speeds or source data jitter.

Not for digital audio, you don't, or you have altered the pitch.

No, hes right, you can resample, but you dont preserve the original data, so its a bit pointless.

True, but you can also use a data-rate converter, which may be what he
meant. I confess that I hadn't even considered the possibility that he
was referring to D/A-A/D resampling, which would indeed be terminally
brain-dead as a jitter suppression technique!
--

Stewart Pinkerton | Music is Art - Audio is Engineering