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Old 20th October 2006, 11:21 AM   #41
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Quote:
Originally posted by BlackCatSound


Please do not confuse jitter and latency.

You've already got YEARS of latency/delay in the recording as it was recorded a long time before you play it.

This 'latency jitter' (apart from being mostly a madeup marketing term) has nothing to do with single track playback. don't get suckered in by jargon.
Unfortunately that is one of the very few explanations I came across, which sound logic to me. I regard it as neutral. RME is
not trying to bring its gear into discussion at that point. Beside that there are known for good drivers. I assume they know what they are talking about.
I'd be happy if somebody would come up with a better explanation.

Anyhow I do not intend to defend that theory.
But one thing I know for sure. With Vista MS is introducing some kind of exclusive mode, which gives you direct access to your outputs to avoid let's call it impact ( and not Jitter/LatencyJitter) by other sources! (Perhaps another marketing trap? Who knows!)
It seems that they are well aware of the related problems.

Back to Linux: I'd call that Vista Exclusive approach realtime process under Unix.
I believe that especially within this area Linux could do a better job.

When it comes to extra jitter on top of the let's call it source data jitter, which is already on your CD, I think it is more than obvious that you'll catch it on the way towards your DAC, starting at the last instance doing some kind of very good buffering/reclocking whatsoever.
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Old 20th October 2006, 11:42 AM   #42
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Jitter is a non-issue in the digital domain as you have no concept of exact timing. The samples are assumed to be a fixed period apart.

Data jitter and clock jitter are totally irrelevant here.

You can process the samples at 1x or 1,000,000x real time, or even process some, wait for a bit, process some more, wait some more and you won't introduce any jitter.

Jitter only ever comes into play when going between the analogue an digital domains. So long as the DAC has a steady supply of data and a good clock it won't matter how you process the data.

If you can't supply data to the DAC in time or have a noisy DAC clock then you get problems.

RME's article is aimed at people doing multi-track, multi layered recordings from multiple analogue and digital sources. In this case varying latency from the different sources does have an impact.
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Old 20th October 2006, 11:44 AM   #43
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Quote:
Originally posted by Bas Horneman

So, what would that implementation look like? Putting a flac file on your usb stick in your pc?

As long as you regard the USB stick as system RAM, just give it a try.

Otherwise.

Either you use full file buffering as it is supplied by foobar
or you install a ramdisc where you copy the track or your full
CD (takes 15s on my PC), for applications not supporting
full file buffering.

These days RAM extensions are quite affordable.
For less than 100$ you get 1GB of quality RAM.
An audio PC IMHO should have anyhow at least 2GB RAM.
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Old 20th October 2006, 11:45 AM   #44
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I didnt see anything in the RME article which actually says that latency jitter has an impact on normal playback quality, only the triggering of synths and samplers. even when it is discussing synth playback it seems only to be in terms of the delay between, for example, a note played on a midi controller keyboard and the point when the triggered sound is heard. There is no suggestion that the triggered sound itself, albeit later than desired, sounds any worse than if it were played extactly on time. Seems the only unique thing about RME's article is that they use the term 'jitter', as lots of places discuss latency.

If you want to try linux - All the efforts in this area in linux are included in the demudi distribution - specifically a low latency patched kernel (audio given priority) and the harmonisation between this and the JACK low latency audio server (you change latency and sample rate settings with this) which connects the audio output of one programme to another (or the soundcard drivers). these things can be set up in nearly any linux system, but come standard in demudi to avoid difficult set up for those who do not want to get into it. this system is built for audio from the ground up. You can get specific latency figures for this distribution based on your hardware, if you know what you can achieve in xp you can make a direct comparison.

I am going ot try your configuration at some point with ramdisk. ramdisk can be set up in linux easily example
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Old 20th October 2006, 11:49 AM   #45
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Quote:
Originally posted by rossco_50
Soundcheck,


There is a very interesting thread at diyhifi on usb asio jitter with measurements and it is considerably worse than spdif - However I know this does not mean it necessarily sounds worse. Would be interesting to see how your optimised setup measures, you may have side stepped the driver problems that were considered to be causing the distortion.
Could you post the link please?


I'll have a look at demudi over the weekend.
Sounds very interesting.
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Old 20th October 2006, 11:50 AM   #46
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Quote:
Either you use full file buffering as it is supplied by foobar
aH..thanks..is that standard in Foobar? Or do I have to set an option.
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Old 20th October 2006, 12:08 PM   #47
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Default USB jitter

usb jitter

There's actually quite a bit of stuff over there on usb dacs, a number of chips have been tested.
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Old 20th October 2006, 12:27 PM   #48
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Quote:
Originally posted by BlackCatSound

Data jitter and clock jitter are totally irrelevant here.

So long as the DAC has a steady supply of data and a good clock it won't matter how you process the data.



I don't agree. What you mess up in the digital domain won't be resolved at a later stage! DAC rule: S..t in = S..t out!
Unless, as mentioned before, you gotta have a very good DAC with a buffer and reclocking stage in front of it!

Further - If that's true what you're saying ouput drivers such as the one from usb-audio.com, which just works in the digital domain, shouldn't have any impact at all!

Again - I am still missing the explanation why changes are happening and are audible, in case where the ouput is considered bit perfect!



Quote:
Originally posted by BlackCatSound


RME's article is aimed at people doing multi-track, multi layered recordings from multiple analogue and digital sources. In this case varying latency from the different sources does have an impact.

I guess I understood what RME with its article is aiming at.
It is just the principle behind it, that buffer contents due to MS interrupt handling are not read and transfered in a linear way.

I still believe you can map that on the audio-path to a certain extent, especially if regarded as a realtime stream.
Though I really do not have a clue how many times the audio datastream is buffered or catching some other unlinarities before leaving the pc-port.

Of course there are more things to consider, when looking at the PC. Such as power supplys, voltage regulation, noise asf..

I was reading that e.g. while running the harddisc the noise level increases in the system.
Of course something like that and more will have an impact on the data signal.
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Old 20th October 2006, 12:34 PM   #49
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Quote:
Originally posted by Bas Horneman

aH..thanks..is that standard in Foobar? Or do I have to set an option.
It's an option under Preferences-Advanced-FullFileBuffering.

The size is to be entererd in kB. Files bigger than the configured
buffer will not be loaded into RAM.
On a 15min track it'll take some seconds to load. That can be a bit annoying if you play a playlist, where breaks between the tracks
are not really wanted.

In this case it would be better to have a ramdisc.
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Old 20th October 2006, 12:58 PM   #50
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Quote:
It's an option under Preferences-Advanced-FullFileBuffering.
Thanks..will try it this evening.
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