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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 25 2008 at 05:29
Well ... last night I found a solution: I installed a different ASIO driver (http://www.asio4all.com), and suddenly all the problems just disappeared. I'm now able to use the lowest buffer setting, resulting in a latency of only 2-4ms ... without any dropouts or synchronisation problems, even with heavy cpu & disk load.Smile

Thanks for all your suggestions ... and if you ever need to use ASIO, check out that driver! It should work with any device which is capable of ASIO.


Edited by MikeEnRegalia - July 25 2008 at 05:30
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Points: 21257 		Direct Link To This Post Posted: July 24 2008 at 08:11
^ I want to decrease latency ... if I increase it everything's fine. I'll try disabling multiprocessor support though. This is usually what you have to do ... make a change and see if there's an improvement, if not then reverse the change and look for something else.
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Certif1ed View Drop Down
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Points: 7559 		Direct Link To This Post Posted: July 24 2008 at 07:20
^Out of interest, what happens if you disable multiprocessor support (assuming your processor is either hyperthreaded or dual core, in which case the mechanisms to disable "multi" processor are different) and increase the latency?
The important thing is not to stop questioning.
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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 24 2008 at 05:33
^ well, I won't go back to XP again ... I guess this simply means that I'll have to live with ~10ms latency for the time being. As long as I can use direct monitoring for recording guitar, it's not that much of a problem anyway ... and keyboards are recorded as midi anyway, and with 10ms latency I can use quantisation to improve the accuracy. Currently I'm just fooling around anyway. Smile
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Points: 7559 		Direct Link To This Post Posted: July 24 2008 at 05:15
^The point is that you're using ASIO, and Vista kludges around it - ASIO/WDM performance is notably worse for latency under Vista than it is under Windows XP.
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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 24 2008 at 04:49
Originally posted by Certif1ed Certif1ed wrote:

Ah - you mentioned Vista.
 
Despite all the claims, most musician friends of mine have found that Vista suffers more from latency than XP did - due to a design flaw in the new WaveRT subsystem, and the fact that WaveRT doesn't work with external audio devices and falls back to an emulator of the old DirectSound (WDM+ASIO support) subsystem that performs somewhat poorly compared to Windows XP.
 
On the Sonar forums, people have mentioned disabling multiprocessor support and UPPING the latency to 15ms (apparently this actually works for some people!), because although Sonar supports WaveRT, the drivers typically don't, and when 7ms or lower has been set, all kinds of glitches appear.
 
 
Note that increasing the buffer does not necessarily increase the latency in Vista.
 
It's improved in Vista SP1 - but not for USB devices, as you'll also note from the MSDN article that the audio engine calls for an exclusive mode event-driven capture, which only works with PCI.
 
 
I'd be happy to be proved wrong, and am interested to see how you get on.
 


Ableton doesn't currently support WaveRT - I use ASIO. The KB37 should arrive soon ... once it's installed I'll install some other software to test it, including Sonar. I'll definitely check out whether WaveRT works.Smile
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Points: 7559 		Direct Link To This Post Posted: July 24 2008 at 04:16

Actually, I wouldn't have thought the requirements were that high, just to run the Toneport - but even so, a P4 1.2 Ghz processor is a very low spec indeed these days.

That spec for the M-Audio is prehistoric - surely no-one attempts to run a music studio on something as pitiful as that? You couldn't run Windows XP on it, that's for certain, let alone a DAW.
 
The M-Audio is in no way a similar device to the TonePort - maybe the soundcard bit is, but the all-important amp-modelling is a million miles away from what the FastTrack does.
 
512Mb RAM is optimistic for any home recording environment - and so is a 1.2Ghz processor.
 
I'm running a dual core 3.2Ghz processor, with 2Gb RAM and a SATAII disk subsystem - and with a >5 minute track, loading and saving projects can take a couple of minutes (which feels like a long time). DAWs are incredibly resource-hungry, and need plenty of RAM, a fast CPU, a well-optimised Operating System, a decent soundcard and above all, a very fast and LARGE disk subsystem. I have 1Tb, and it's beginning to feel too small.
 
The Line6 software isn't noticeably resource hungry, though - I'll take a look at the Mem/CPU usage when I fire up "the studio" tonigh.
 
 


Edited by Certif1ed - July 24 2008 at 04:19
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Vompatti View Drop Down
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Points: 67421 		Direct Link To This Post Posted: July 24 2008 at 03:58
Are the system requirements for the Toneport really as high as stated? For the Toneport UX1 they are:

Pentium IV 1.2GHz or better (2.5GHz or more recommended)
512MB RAM minimum (1GB or more recommended)

while a similar devide, M-Audio Fasttrack USB only requires a Pentium II 350 w/ 64MB RAM. Is this because of the software that comes with the Toneport or what?
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Points: 7559 		Direct Link To This Post Posted: July 24 2008 at 03:36
Ah - you mentioned Vista.
 
Despite all the claims, most musician friends of mine have found that Vista suffers more from latency than XP did - due to a design flaw in the new WaveRT subsystem, and the fact that WaveRT doesn't work with external audio devices and falls back to an emulator of the old DirectSound (WDM+ASIO support) subsystem that performs somewhat poorly compared to Windows XP.
 
On the Sonar forums, people have mentioned disabling multiprocessor support and UPPING the latency to 15ms (apparently this actually works for some people!), because although Sonar supports WaveRT, the drivers typically don't, and when 7ms or lower has been set, all kinds of glitches appear.
 
 
Note that increasing the buffer does not necessarily increase the latency in Vista.
 
It's improved in Vista SP1 - but not for USB devices, as you'll also note from the MSDN article that the audio engine calls for an exclusive mode event-driven capture, which only works with PCI.
 
 
I'd be happy to be proved wrong, and am interested to see how you get on.
 


Edited by Certif1ed - July 24 2008 at 03:50
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Points: 21257 		Direct Link To This Post Posted: July 24 2008 at 03:04
My computer is 64bit ... but that has nothing to do with performance. It's simply a different architecture - the main difference is that you can use more memory (I have 4GB).

Thanks for bothering though - I really appreciate the effort. I know the Ableton FAQ section, and the bit you posted is not related to my problem, unfortunately.
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Points: 415 		Direct Link To This Post Posted: July 23 2008 at 07:19

Yes, but I think that's the crux of your problem- it should on a 64bit machine. I take it you have at least 2GB of memory? Assuming Ableton is Vista ready and is a 64bit installation package, it sort of rules out the software and the computer hardware and can only point to the line 6 unit. Does that make sense, or am I jumping the gun somewhere?

addition: You have probably already seen this but it seems to have relevance.

from the FAQ's section in Ableton
When do I need to adjust the "Overall Latency" setting in Live?
Suppose you are monitoring your voice or an instrument through an external device instead of Live (you have the "Monitor through Live" switch set to "no"). Now, as you sing or play along with the metronome or clips already in the Set, you are in fact playing along with "late" audio, because it takes some milliseconds for the audio to travel from the CPU to the audio outputs. This delay is referred to as "latency." Because you are hearing the audio from the program late, the audio you are recording into the program arrives late with respect to the program's internal timing. Live automatically compensates for this error by moving the recorded audio in song time so that you normally don't have to worry about any of this. If the recording you make sounds "untight" despite Live's effort to compensate for latency, then the audio hardware driver is probably the culprit: It is reporting an inaccurate latency value to Live. You can manually correct Live's latency assumption by adjusting the "Overall Latency" control in the Audio Preferences. The manual describes how to do this in the "Computer Audio Issues" chapter.



Edited by cobb2 - July 23 2008 at 07:50
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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 23 2008 at 06:44
^ I think I mentioned before that the lowest buffer setting doesn't work. Why else would I have created this thread? Wink
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cobb2 View Drop Down
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Points: 415 		Direct Link To This Post Posted: July 23 2008 at 06:28

If you have a 64bit CPU you should be able to set your buffer to its lowest setting. Does Apleton support 64bit?

I think Vista has a different audio subsystem than XP- though this shouldn't affect you if you are using ASIO



Edited by cobb2 - July 23 2008 at 06:30
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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 23 2008 at 06:24
I'm still waiting for the KB37 to arrive ... if all else fails, maybe I'll write them an email, but they don't yet officially support Vista 64bit, so I guess that for the time being I'll have to find a solution myself.

One possible solution would be to decrease the buffer size while recording the midi instruments ... there would be slight errors in the playback, but it wouldn't affect the recorded midi information. Still, I'd like to have an environment where I can focus on playing/recording the music and not concern myself with buffer settings and hardware configurations all the time. After all, that's why people use DAWs in the first place ... because it's much more comfortable and efficient than using real studio hardware.
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cobb2 View Drop Down
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Points: 415 		Direct Link To This Post Posted: July 23 2008 at 06:10

Oh, okay- just bandying ideas around. From experience I know that it can very frustrating getting music recording software to play nice with audio hardware.

addition: have you emailed Albeton with the problem- if they see you as a potential customer, I'm sure they would be more than happy to help



Edited by cobb2 - July 23 2008 at 06:12
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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 23 2008 at 05:55
Of course a larger buffer works ... but larger buffer means higher latency, which is my problem.
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cobb2 View Drop Down
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Points: 415 		Direct Link To This Post Posted: July 23 2008 at 05:35

Did you fix your problem? What about your buffer size? Have you played with this? I think it works the smaller the buffer size the quicker the actual computer needs to be to process it in real time- so a larger buffer may work. (I think that is the right way around).

edit: read quicker computer as more powerful



Edited by cobb2 - July 23 2008 at 05:36
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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 23 2008 at 04:52
^ the Line6 interface does that automatically ... that's not the point of this thread. The point is that direct monitoring is not possible when you want to record drums, keyboards etc. entirely within the DAW application, using a MIDI keyboard as input.
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Points: 8844 		Direct Link To This Post Posted: July 22 2008 at 16:21
I avoid the latency issue altogether.  I put a splitter into my output and send half of my signal to the computer and half to an amp.  

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MikeEnRegalia View Drop Down
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Points: 21257 		Direct Link To This Post Posted: July 22 2008 at 09:49
^ Maybe that would improve the latency, but I really want to record in 24bit/96khz. It's the standard for professional audio, and even if it gets downsampled in the end the high resolution is necessary during the production process. Most DAWs do their calculation in this resolution anyway, and using lower quality samples for recording only results in more down/upsampling calculations.
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