On Tue, 2 Dec 2003 16:29:41 +0000, Ian Molton wrote:

On Tue, 2 Dec 2003 16:18:34 +0000 (UTC)
(Stewart Pinkerton) wrote:

For that, you require to reclock
the datastream from an independent free-running low-noise clock, and
this is *very* rare.

the important detail being that unless you have BI-DIRECTIONAL data transfer you can never reliably re-clock the data, unless you have *gargantuan* buffers (mind you with todays memory prices...)

the point is that in a continuous stream, sooner or later you will have a problem as the free running local DAC wont be running at the same speed as the data stream, so it will, eventually either over- or under- run its data buffer and have either some skipped or lost data, or some 'dodgy' compensation scheme.

whats needed to do it *right* is a data stream that can deliver the data *faster* than its needed, and a source that can pause its transfer.

then the source can transfer at full speed to the DACs buffers, and the DAC can say 'no more please I'm full'.

then as the buffer drains the DAC can request more data.

this utterly eliminates jitter and frankly Im amazed the high-end HiFi audio industry hasnt cottoned on to the idea yet, if not because its audibly better, but because it means they can sell a whole load more interconnects and new DACs etc.

Too late - see the Meridian 800 series.
--

Stewart Pinkerton | Music is Art - Audio is Engineering

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

Ian Molton wrote:

On Tue, 02 Dec 2003 19:36:53 +0000
Ian Bell wrote:

I am well aware of the term slaved and if you had read my post more
closely
you would have realised I was *not* referring to a slaved clock.

You cant feed SPDIF to a non-slaved clock. if you do you will need either
short tracks, or massive buffers, and you better *pray* the spdif clock is
marginally faster than the free-running one.


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.
--

Stewart Pinkerton | Music is Art - Audio is Engineering

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

Ian Molton wrote:

On Tue, 02 Dec 2003 19:36:53 +0000
Ian Bell wrote:

I am well aware of the term slaved and if you had read my post more
closely
you would have realised I was *not* referring to a slaved clock.

You cant feed SPDIF to a non-slaved clock. if you do you will need either
short tracks, or massive buffers, and you better *pray* the spdif clock is
marginally faster than the free-running one.


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.
--

Stewart Pinkerton | Music is Art - Audio is Engineering

On Tue, 2 Dec 2003 20:00:14 +0000, Ian Molton wrote:

On Tue, 02 Dec 2003 09:15:24 +0000 (GMT)
Jim Lesurf wrote:

TBH though I recon USB audio would be a decent alternative to all this
unidirectional crap ;-)

I have the feeling that we have had this discussion before, elsewhere...
:-)

Yeah I was thinking that ;-)

One thing we never did resolve though...

Why *is* audio digital transport done in a way that has no feedback?

Because the people who designed the system never thought that anyone
would be dumb enough to split the transport and DAC sections, or if
they did, that they wouldn't use a single master clock in the DAC to
slave the transport. Then along came the notoriously incompetent
so-called 'high end' audio industry.........
--

Stewart Pinkerton | Music is Art - Audio is Engineering

On Tue, 2 Dec 2003 20:00:14 +0000, Ian Molton wrote:

On Tue, 02 Dec 2003 09:15:24 +0000 (GMT)
Jim Lesurf wrote:

TBH though I recon USB audio would be a decent alternative to all this
unidirectional crap ;-)

I have the feeling that we have had this discussion before, elsewhere...
:-)

Yeah I was thinking that ;-)

One thing we never did resolve though...

Why *is* audio digital transport done in a way that has no feedback?

Because the people who designed the system never thought that anyone
would be dumb enough to split the transport and DAC sections, or if
they did, that they wouldn't use a single master clock in the DAC to
slave the transport. Then along came the notoriously incompetent
so-called 'high end' audio industry.........
--

Stewart Pinkerton | Music is Art - Audio is Engineering

"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.

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.

"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.

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.

On Tue, 2 Dec 2003 22:51:17 +0000 (UTC)
(Stewart Pinkerton) wrote:

Then along came the notoriously incompetent
so-called 'high end' audio industry.........

The rest is history, huh?

--
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 22:51:17 +0000 (UTC)
(Stewart Pinkerton) wrote:

Then along came the notoriously incompetent
so-called 'high end' audio industry.........

The rest is history, huh?

--
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, 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.