How to get ASIO working in SFP 4?
Okay, I remember a long time ago looking through these forums and seeing just how difficult it is to find some manner of tutorial to get ASIO running in SFP 4 (not in XTC mode as XTC mode works perfect for me now). I remember finding a FAQ or something like that, but after 5 hours of searching through the forums today have come up empty.
Basically, how do you set up Scope 4 to send ASIO I/O to your sequencer? I know you have a Wave module running, but that's about the extent of it. I can set ASIO IO in Sonar now, but obviously I get no audio on anything I put into the system.
My basic structure for audio is now setup as:
All audio in through to A16-Ultra, then through Z-Link A and B to Scope Pro. I have the studio monitors plugged into the first two outs of the A16 Ultra.
The basic setup in Scope is the 1632 mixer, with the Scope Z-Link A with all audio going to individual lines on the mixer, and all audio out going through the two mixed line outs to the Scope Z-Link A out on the first two channels. Works just for playing instruments and using the mics, and that's all.
One point of note is that Sonar (at least 3.1.1 which is what I use) does not support ASIO 2.0 or above, so I'm stuck to 1. XTC set to Asio 1 works flawlessly in Sonar.
Any help in this regards is greatly appreciated. Even a setup file that someone has made would help me to understand how ASIO is supposed to be routed, etc.. I mean, why does it require the WAVE module to be loaded, just to work, even though I don't even have the WAVE module actually linked to anything on the 1632 mixer and I still get default computer audio routing through it without problems?
Be well,
-Tak
Basically, how do you set up Scope 4 to send ASIO I/O to your sequencer? I know you have a Wave module running, but that's about the extent of it. I can set ASIO IO in Sonar now, but obviously I get no audio on anything I put into the system.
My basic structure for audio is now setup as:
All audio in through to A16-Ultra, then through Z-Link A and B to Scope Pro. I have the studio monitors plugged into the first two outs of the A16 Ultra.
The basic setup in Scope is the 1632 mixer, with the Scope Z-Link A with all audio going to individual lines on the mixer, and all audio out going through the two mixed line outs to the Scope Z-Link A out on the first two channels. Works just for playing instruments and using the mics, and that's all.
One point of note is that Sonar (at least 3.1.1 which is what I use) does not support ASIO 2.0 or above, so I'm stuck to 1. XTC set to Asio 1 works flawlessly in Sonar.
Any help in this regards is greatly appreciated. Even a setup file that someone has made would help me to understand how ASIO is supposed to be routed, etc.. I mean, why does it require the WAVE module to be loaded, just to work, even though I don't even have the WAVE module actually linked to anything on the 1632 mixer and I still get default computer audio routing through it without problems?
Be well,
-Tak
You don't need the Wave driver loaded to have it work, the Wave Source/Wave Dest are for the normal DirectSound/MME Windows stuff, they're not needed for ASIO, ZLink, ADAT or Analog signals to work.
As for the specific ASIO question, the modules you are looking for are in the "Software IOs" submenu. You will need one ASIO Source and one ASIO Dest module. The ASIO Source are the channels coming *from* the sequencer, while the ASIO Dest are the channels that go *to* your sequencer. As you are running Sonar 3, you will want to use the ASIO/ASIO1 modules, and not the ASIO2 modules
There's a few variation of each module, depending on wether you want 16 bit, 24 bit, 32 bit, or 32 bit float connections to your sequencer. The "IOs & Drivers" section of the SFP manual has some more information about each type of modules.
To set the number of available ASIO channels, just double-click on the ASIO Source module, and it'll let you configure how many Source and Destination Channels will be available to your sequencer. The normal ASIO modules support up to 32 (mono) channels, while those branded with a 64 (i.e. ASIO1-16 Source 64) support up to 64.
On the sequencer-side, I don't know enough about Sonar to help you much, but the first thing you need to do is set the Sonar audio driver to Scope ASIO. Then you can configure your audio channels to listen/record the different ASIO channels coming in from Scope.
As for the specific ASIO question, the modules you are looking for are in the "Software IOs" submenu. You will need one ASIO Source and one ASIO Dest module. The ASIO Source are the channels coming *from* the sequencer, while the ASIO Dest are the channels that go *to* your sequencer. As you are running Sonar 3, you will want to use the ASIO/ASIO1 modules, and not the ASIO2 modules
There's a few variation of each module, depending on wether you want 16 bit, 24 bit, 32 bit, or 32 bit float connections to your sequencer. The "IOs & Drivers" section of the SFP manual has some more information about each type of modules.
To set the number of available ASIO channels, just double-click on the ASIO Source module, and it'll let you configure how many Source and Destination Channels will be available to your sequencer. The normal ASIO modules support up to 32 (mono) channels, while those branded with a 64 (i.e. ASIO1-16 Source 64) support up to 64.
On the sequencer-side, I don't know enough about Sonar to help you much, but the first thing you need to do is set the Sonar audio driver to Scope ASIO. Then you can configure your audio channels to listen/record the different ASIO channels coming in from Scope.
symply just open the "Asio source&destination" connect the source to the mixer inputs and the destination to the L-R outputs and that's it.....
if you need more then one 2 inputs and outputs just double click on the asio source and destination and change in the new screen how many in/out you need and connect them again......
have fun....
and by the way.... if you using sonar you need also to open the "Wave source" and to connect it to the mixer..... to hear some wav files....
if you need more then one 2 inputs and outputs just double click on the asio source and destination and change in the new screen how many in/out you need and connect them again......
have fun....
and by the way.... if you using sonar you need also to open the "Wave source" and to connect it to the mixer..... to hear some wav files....
Thanks very much for the help you two. But my troubles continue, sadly. I've gotten ASIO to be recognized and basically working in Sonar, BUT...
Now I've introduced a LOT of crackles and whistles and pops.. But it's very weird, because it only happens when I add effects AND audio in Sonar. AS I'm playing with full effects running, I get no problems, but if I record it then I get all the noise, but only as I'm playing back..
I've tested with both putting the effects in line with the audio coming straight from the A16 Ultra, AND with just putting the effects in after recording plugging them in after the ASIO Source module.
Either way, if effects are played I get horrible noise. The only thing I can think of is that the ASIO format between Sonar and SFP is somehow different introducing bad problems. (for reference, if I use no effects at all, I usually don't hear any problems, though I did hear a couple of small pops and the like every now and then, which is weird because it's totally random, and only after recording, which means it's solely on the playback side of the system.)
I'm going to search through the forums here and over at Cakewalk for further help on this, but I'm also going to test it in XTC mode to see if the same noises are encountered when running it through effects.
Has anyone had problems with the wrong ASIO settings in sonar? i.e. bit depth and frequency, and the like.. I *THINK* I have it set correctly, but I can't be sure.
Now I've introduced a LOT of crackles and whistles and pops.. But it's very weird, because it only happens when I add effects AND audio in Sonar. AS I'm playing with full effects running, I get no problems, but if I record it then I get all the noise, but only as I'm playing back..
I've tested with both putting the effects in line with the audio coming straight from the A16 Ultra, AND with just putting the effects in after recording plugging them in after the ASIO Source module.
Either way, if effects are played I get horrible noise. The only thing I can think of is that the ASIO format between Sonar and SFP is somehow different introducing bad problems. (for reference, if I use no effects at all, I usually don't hear any problems, though I did hear a couple of small pops and the like every now and then, which is weird because it's totally random, and only after recording, which means it's solely on the playback side of the system.)
I'm going to search through the forums here and over at Cakewalk for further help on this, but I'm also going to test it in XTC mode to see if the same noises are encountered when running it through effects.
Has anyone had problems with the wrong ASIO settings in sonar? i.e. bit depth and frequency, and the like.. I *THINK* I have it set correctly, but I can't be sure.
Well with Sonar in mind, what ASIO module should I use? (assuming 24/96 audio) I've tested a few - obviously the ASIO1 modules only. I was just about to use the ASIO 24 module, before I had to leave for lunch.
Would you like a screenshot of the settings I have set in Scope 4?
And by changing things in Sonar, the only thing I meant was changing the bit depth, and the sample rate.. obviously I want/need 24/96.
Would you like a screenshot of the settings I have set in Scope 4?
And by changing things in Sonar, the only thing I meant was changing the bit depth, and the sample rate.. obviously I want/need 24/96.
Okay, I hate regressing. Now, no matter what modules I use, suddenly Sonar no longer is recording any audio. I've got no errors reported anywhere, and I've tried most of the ASIO 1 modules, as well as 2 of the ASIO 2 modules, each one always set to DL1 and DR1 from IL1 and IR1. (Scope Z-Link A channels 1 and 2 go to IL1 and IR2).
My father mentioned something about changing the send on each channel, perhaps? maybe I have IL1 and IR2 routed to a different set of channels. Though because I don't know how to do that, I don't know how it would have happened in the first place. To verify this, I'm going to put the ASIO Dest module connected to the Mix L and R.
but garyb: do you have a .pro file of the modules and setup you use for Sonar and Scope 4? I'm going nuts just trying to figure out what I'm missing.
Thanks, be well.
-Tak
My father mentioned something about changing the send on each channel, perhaps? maybe I have IL1 and IR2 routed to a different set of channels. Though because I don't know how to do that, I don't know how it would have happened in the first place. To verify this, I'm going to put the ASIO Dest module connected to the Mix L and R.
but garyb: do you have a .pro file of the modules and setup you use for Sonar and Scope 4? I'm going nuts just trying to figure out what I'm missing.
Thanks, be well.
-Tak
Okay, further testing ensues. I've managed to all but eliminate the pops and crackles by simply changing the ULLI up one level (from 1ms to 2ms under 96khz). However, They happen constantly when the Wave Source is connected and a wave file is playing. I haven't tried all the way up to the worst ULLI setting, so I will try that next, but it is VERY bad when wav files play.
Is there perhaps a better module for me to use for .wav files? i tried the 24 bit wave source and got no audio out of it at all.. am I doing something wrong?
Is there perhaps a better module for me to use for .wav files? i tried the 24 bit wave source and got no audio out of it at all.. am I doing something wrong?
You should stick to properly setting up ASIO first, forget about the Wave Source/Dest playing stuff. When ASIO works well, you can do the Wave Source/Dest debugging, otherwise you might end up ping-ponging back and forth between the too.
As for ULLI stuff, you should try setting it up at the highest value, and slowly bring it up, testing each value to see where the pops and crackles begin. Test it out with a fairly heavy project to make sure it'll perform well under heavy load.
As for ULLI stuff, you should try setting it up at the highest value, and slowly bring it up, testing each value to see where the pops and crackles begin. Test it out with a fairly heavy project to make sure it'll perform well under heavy load.
Therein lies the problem. I've been trying to figure out how to properly setup ASIO on this system. It SEEMS correct, but logically it's not right (at least in the way I understand it). I'm currently loading ASIO 2 32bit source/dest modules in Scope 4 and in Sonar, I've got it selecting them as normal (though Sonar doesn't supposedly support ASIO 2.0). But things seem to be working. I have ULLI setup on the third setting from the top, and I don't get any crackles or noise, etc, from anything ASIO, so I'm *thinking* it's working well. Now when I add the Wave Source module, I suddenly get cracks and pops from the computers default sounds.On 2005-07-25 15:00, symbiote wrote:
You should stick to properly setting up ASIO first, forget about the Wave Source/Dest playing stuff. When ASIO works well, you can do the Wave Source/Dest debugging, otherwise you might end up ping-ponging back and forth between the too.
I kind of went the exact opposite direction, using the fastest ULLI setting, then slowing it down until I couldn't get any crackles and pops. BUT, I don't have any setups for a heavy load. It's something my father would know more about, as I'm more of a computer technician, and he's the musician.As for ULLI stuff, you should try setting it up at the highest value, and slowly bring it up, testing each value to see where the pops and crackles begin. Test it out with a fairly heavy project to make sure it'll perform well under heavy load.
Does ANYONE have a good example .pro file of an ASIO setup for sequencers, preferably for Sonar? Anything with a heavy load, as is described, etc.
Thanks,
-Tak
1-3ms is only really necessary for when you're recording and needing to setup realtime monitoring through your app (software). With the routing possibilities of scope software monitoring becomes far less important (one reason many people use ASIO1 still).
3-7ms should be fine for most normal compositional & editing tasks, and raising the buffer size up to 13 or even 23ms will reduce cpu load when you're nearing the end of a project (I find myself working at 7 or 13ms mostly).
For the WAV driver, go into your control panels, open the "Sounds and Audio Devices" control panel, go to the "Hardware" tab and find (one of) your scope card(s), either double click it or choose properties. When the "Creamware (blabla) Properties" panel is open, go to the "Settings" tab and adjust the "Output preload" slider under Wave Setup. Note that the 'base' latency of the wav driver is based of off ULLI but the 'output preload' setting seems to do something along the lines of changing the 'number of buffers' settings in Wavelab / Cubase MME.
3-7ms should be fine for most normal compositional & editing tasks, and raising the buffer size up to 13 or even 23ms will reduce cpu load when you're nearing the end of a project (I find myself working at 7 or 13ms mostly).
For the WAV driver, go into your control panels, open the "Sounds and Audio Devices" control panel, go to the "Hardware" tab and find (one of) your scope card(s), either double click it or choose properties. When the "Creamware (blabla) Properties" panel is open, go to the "Settings" tab and adjust the "Output preload" slider under Wave Setup. Note that the 'base' latency of the wav driver is based of off ULLI but the 'output preload' setting seems to do something along the lines of changing the 'number of buffers' settings in Wavelab / Cubase MME.
if you don't use vst instruments, than you can set ulli to 25ms and you will still hear everithing in realtime (zero latency).
so, ulli is important only while playing vst instruments (if your project is good) and 7ms will be ok.
btw: i still use asio1.
_________________
<font size=-2>i got my mojo working, but it just won't work on you</font>
<font size=-1>[ This Message was edited by: sandrob on 2005-08-01 00:24 ]</font>
so, ulli is important only while playing vst instruments (if your project is good) and 7ms will be ok.
btw: i still use asio1.
_________________
<font size=-2>i got my mojo working, but it just won't work on you</font>
<font size=-1>[ This Message was edited by: sandrob on 2005-08-01 00:24 ]</font>
a couple quick points: 1), I've tested all the settings and haven't really noticed the time lag and all, so the actual time lag is only really noticeable by my father. I'll have to verify whether or not it's necessary to have the highest ULLI (though with the studio we have, it SHOULD be able to have that setting). To me it is only really noticeable when I record both midi and audio simultaneously. I can then hear an extremely slight lag time.
2) we do all our recording at 24/96, so even the higher latency settings are still lower than what everyone is mentioning (which seems to be for either 44.1khz or 48khz).
valis: thanks for the info on the wav driver. That shall come in handy I think. As far as the other settings, I think it's more just an aesthetic issue with my father, whether or not he wants the extremely low latency.
sandrob: How would I hear everything in real time without VST instruments? I mean I'm not using them, just straight audio in and out of Sonar, and I can still tell that there is some very low latency (based on the midi and audio recording simultaneously).
garyb: that's a good suggestion about setting priorities. I'll have to try it to test if SFP really doesn't need that high of a priority while running with audio. I'd have imagined that it needed higher priorities, but I suppose once settings are set, the unit does all the work on the Scope card.
2) we do all our recording at 24/96, so even the higher latency settings are still lower than what everyone is mentioning (which seems to be for either 44.1khz or 48khz).
valis: thanks for the info on the wav driver. That shall come in handy I think. As far as the other settings, I think it's more just an aesthetic issue with my father, whether or not he wants the extremely low latency.
sandrob: How would I hear everything in real time without VST instruments? I mean I'm not using them, just straight audio in and out of Sonar, and I can still tell that there is some very low latency (based on the midi and audio recording simultaneously).
garyb: that's a good suggestion about setting priorities. I'll have to try it to test if SFP really doesn't need that high of a priority while running with audio. I'd have imagined that it needed higher priorities, but I suppose once settings are set, the unit does all the work on the Scope card.
24/96 sounds better but is a complete waste of resources imho, especially if you are going to downsample to 44.1......
my suggestion, work at 24/44.1 or just know that you'll not be able to do a whole lot with your computer, as 96k will eat resources, both processing and storage like a pig....
96k is a marketing tool for hardware manufacturers(unless your final result will be at 96k). imho, downsampling does more damage to a signal than using a lower sample rate. do a search for 96k for all the contentious arguments for and against. most REAL money productions are done at 44.1 or 48k.......
my suggestion, work at 24/44.1 or just know that you'll not be able to do a whole lot with your computer, as 96k will eat resources, both processing and storage like a pig....
96k is a marketing tool for hardware manufacturers(unless your final result will be at 96k). imho, downsampling does more damage to a signal than using a lower sample rate. do a search for 96k for all the contentious arguments for and against. most REAL money productions are done at 44.1 or 48k.......
Well my father is capable of telling the difference between 44.1 and 48k, so all work he does is at a BASE of 48k (no less, unless output to CD).On 2005-08-01 20:03, garyb wrote:
24/96 sounds better but is a complete waste of resources imho, especially if you are going to downsample to 44.1......
my suggestion, work at 24/44.1 or just know that you'll not be able to do a whole lot with your computer, as 96k will eat resources, both processing and storage like a pig....
Processing may be an issue, as we're running only on a 400Mhz FSB with dual Xenon 2.4Ghz processors (I think it's the FSB mostly that's the problem). But storage isn't even slightly an issue as we have a rather nice rackmounted raid setup for Video as well as Audio and the audio portion is substantially smaller than Video even at 24/96 rates.
What's your thoughts on our processing power here?
considering the math, I don't see how downsampling does more damage than simply recording at a certain rate. Most of my work has been done in pictures and video, and the general rule is ALWAYS work at a higher resolution while working with any kind of effects, or anything that would otherwise change or degrade the quality of the original, then downsample. It's kind of like saving an MP3, then saving it as an MP3 again. each successive time you do so it "downsamples" itself, simply by being a lossy codec.96k is a marketing tool for hardware manufacturers(unless your final result will be at 96k). imho, downsampling does more damage to a signal than using a lower sample rate. do a search for 96k for all the contentious arguments for and against. most REAL money productions are done at 44.1 or 48k.......
BUT, I do have to grant, my experience in sound design is VERY small, so when it comes to this system, I'm still learning a lot, and you may very well be absolutely correct. Just doesn't seem that way if Video has anything to say about it
