need an ASIO guru ...

[Gary_101]

hi folks,

just getting back into using my Pulsars after years away from the scene.

I used to use Sonar, which was great, in terms of ease of assigning wave drivers to audio tracks.

Now I've got a copy of Cubase LE, relatively new 2009, and the "asio" thing is driving me up the wall. I always avoided cubase because the "asio thing" seemed convoluted, but now I have no choice.

so, in my Scope project, I've loaded the 'basic' ASIO in's and out's (there's a whole lot of other "asio" choices -> but again, too convoluted. what does it all mean, anyway?) Each ASIO module has a total of 4 channels.

So, I'm using my 2 analog inputs on my P-I card, same on my P-II card, for a total of 4 mono inputs, for live, miked instruments; so PULSAR ANALOG SOURCE & PULSAR 2 ANALOG SOURCE are hooked up to the 4 terminals on the ASIO DEST. module.

in the Cubase LE "Device Setup" box, my second set of ASIO INPUTS shows up as "inactive", and I can't seem to change this. I don't have a clue how "inactive" got there. The rest, in & out's are all "active". So, I try the "asio fit 64" modules, and at least these all show up as "active". Yes, I'm re-starting Cubase LE after every I-O module change in SCOPE.

So, I assign 'mono' ASIO modules to my live instruments; I know there's signal coming thru, because I tested the P1 & P2 analog sources by routing them into the big mixer; they ARE live.

but when I re-route them thru the "asio fit dest 64" module, no live signal of any kind shows up in Cubase. (SAME with the 'basic' ASIO modules).


Sorry, it's late at night and this Cubase thing is simply pissing me off.

any help or scotch on the rocks would be appreciated.

gary

[babaorum]

In cubase you must verify to have these connections 'inputs' and 'outputs' in 'Devices/VST connections' to have something like that :

But with four inputs in your case.

[garyb]

the asio modules really aren't that much different than the wave modules. you should use ASIO with every sequencer because they work well. you do need a proper pair. ASIO source with ASIO dest or ASIO2 source with ASIO2 dest.

for Cubase, i usually use the ASIO2 flt source, because Cubase can use the Floating point math used with that driver. Cubase can also use the Fixed point math of the ASIO2 32 source. both modules use the same ASIO2 destination module. you can use ASIO instead of ASIO2, i use the later version.

in Cubase, you must obviously choose the ASIO SCope driver. as the previous post shows, you must add stereo busses in VST Connections to add i/o to the program and use all of the available ports in your routing window. this is not an ASIO thing, it's just how cubase works. other programs just make all avialable i/o active upon opening the program. go with it.

there is a Scoperise article that covers setting up Scope with Cubse here:
http://www.hitfoundry.com/issue_01/cubase.htm
you will need to adat the information to fit what you are doing, for how you want to work.

[Gary_101]

thanks for your help guys;

I think I had all the bases covered; I'll post my connections below,
in case I'm missing something. Still no audio shows up. Cubase, I believe,
has a little VU meter, on the right end of the Transport module, but nothing
shows there, and when I hit record, it's a blank track laid down. Again,
I checked Analog in from both Pulsar cards, and both are "live" in the big mixer.
So, somehow, audio is not getting thru to Cubase.

I tried one more variation: used the ASIO2 DEST 64 modules;
this time I am getting some sound through, but only the right-panned mono side of the analog stereo pair inputs. Also, Cubase crapped out trying to record 3 mono tracks simultaneously, which is pretty lousy. Sonar never had this type of an issue with only 3 tracks.

PULSAR WIRING ...


ASIO CUBASE ...


VST CONNECTIONS ...



CUBASE TRACKS ...

[babaorum]

With this project you can record only the analog input of the cards.

If you want record the mix of cubase you must add STM out -> ASIO Dest

Could you show what you have for inputs in device/VST connections ?

[garyb]

a couple of things...

why are you not recording the bus outs or the direct outs of the mixer? why monitor through the sequencer? that makes latency in your monitoring system. on addition, you have no meters the way you are doing things. it's hard to tell why sound is and isn't.

the sequencer is a multitrack recorder. it has jacks to go in and out of it, just like a hardware recorder does. if you had a real studio with a real mixerm even if the recorder had a mixer, you would still run the tracks through the real mixer for mixdown. all monitoring of all signals in the studio would go through the studio's mixer not the recorder's mixer. the Scope routing window is the studio...

the reason there's no sound though, is that you are using an ASIO2 source module and an ASIO1 destination module. read the actual names on the modules and you'll see they don't match. remember i told you that you must have a matching pair. the correct destination module to match the source module is named ASIO2 Dest-64

also, you should use stereo busses in Cubase. a mono bus won't work for a stereo channel. a stereo bus will work fine for a mono channel, you only need assign right or left. using a stereo bus simplifies things. mono busses make complications...

[Gary_101]

Thanks Baya & Gary for your input once again.
Yes Gary, I caught that ASIO-2 error after I did
the screen capture and it didn't make any difference;
or perhaps it did give me the "one-sided mono" effect.

What does seem to work however, in terms of giving me
individual "mono" inputs is configuring both ASIO in's and out's
connections in STEREO, rather than MONO, as you recommended.
So, now all my inputs are working.

Regarding routing inputs thru the mixer. I've always used my 4 available
analog inputs on my 2 Pulsar cards to record 3 to 4 live instruments/voices
in mono, straight to sequencer. Then, once all tracks, perhaps with a couple of
additional MIDI's, are laid down in Sonar or whatever, I PLAY them thru the big
mixer, with FX sends, EQ, compression etc for the final mixdown, ie mastering to
a single stereo track. This is something I just figured out logically on my own,
not having a professional model to go by.

So, you're saying, run the analog "in's" thru the mixer, straight to BUS OUT'S,
and THEN on to the sequencer via ASIO or WAV ...? Is this for checking VU
levels, gating, compressing etc?

One thing of concern that comes up, re, using CUBASE, is that as soon as I
task-switch from CUBASE TO SCOPE, CUBASE stops playing my tracks, so I
can't monitor - say, solo tracks - and adjust FX / EQ etc in real 'play' time.
This is kind of bizarre. I've never seen any audio app NOT PLAY in the background;
Cubase just switches off when I move to SCOPE.
Has anyone else known this? Is there a fix for it?

Thanks to all for you patience with my amateurish fitting of square pegs
into round holes!!

Gary

(ps Baya, my inputs were also mono, so I changed both IN & OUT to stereo, works great)

(PPS - another odd thing about Cubase's MIDI controls; other sequencers
seem to have the ability to "tune" your MIDI in numbered half-tones up or down,
but I can't find any hint of this in CUBASE.)

[garyb]

So, you're saying, run the analog "in's" thru the mixer, straight to BUS OUT'S,
and THEN on to the sequencer via ASIO or WAV ...? Is this for checking VU
levels, gating, compressing etc?

yes, for these reasons(although most of the time i'd rather gate, compress and eq after recording) and also because when you also monitor playback through the mixer, overdubs are in realtime, with ZERO monitoring latency. also, that allows you an easier way to reroute.

One thing of concern that comes up, re, using CUBASE, is that as soon as I
task-switch from CUBASE TO SCOPE, CUBASE stops playing my tracks

there is a setting in the vst setup window, in the "vst audio system" section called "release audio driver when application is in the backround". uncheck the box.

(PPS - another odd thing about Cubase's MIDI controls; other sequencers
seem to have the ability to "tune" your MIDI in numbered half-tones up or down,
but I can't find any hint of this in CUBASE.)

if it can be found in another sequencer, it's in Cubase. in the "inspector", the section to the left of the tracks you can set this up. highlight the midi track that you want to adjust. in the inspector, click on the midi insert section. left click on a slot and choose "midi modifiers". done. this is where to put a quantizer on a midi trach as well.

[babaorum]

(PPS - another odd thing about Cubase's MIDI controls; other sequencers
seem to have the ability to "tune" your MIDI in numbered half-tones up or down,
but I can't find any hint of this in CUBASE.)

If you don't see it in the track inspector, right-click in the track inspector area and check 'Midi parameters', you have ability to tune your midi track into.

[Gary_101]

thanks again for your excellent responses;
looks like I'm finally getting the wrinkles out of ol' Cubase.

would like to know a little more about the "how to's" in terms of
routing live input thru the mixer buses en route to sequencer,
but will take time to formulate that later; look like the key
there is choosing the BUS PAGE view on the STM 2448 mixer
and checking the "blue square" in the diagram that represents
the routing. Back to that later.

One problem I'm noting on mixdown of my sequencer tracks
(at this point, 3 mono WAV and 1 stereo internal MIDI)... I
route the mixing back thru the STM 2448 add fx etc, then send
thru MxL and MxR to both Analog Dest (for monitoring) and
Wave Dest 1 (for mixdown recording).

I have always done my mixdown to stereo in a seperate app,
noteably Sound Forge or Wavelab. I'm now using Wavelab or
Audacity. In either of those 2 apps, there appears a lot of
crackling and popping in my final stereo mix. This is NOT audible
in monitoring the mixdown, but only in the recorded result,
in both apps.

when I google it, it seems to be all about "buffer size", but
what buffer, and where is this adjusted? The only setting
I can see that might be involved is in hardware device manager
for PULSAR 1/2, where, just under "number of devices",there's
an "output preload" setting, a slider between "big" and "small"
which is currently in the middle.

am I on the right track?

THANKS AGAIN, you guys are terrific!~
(it's really a blessing that this community is still running strong after all these years...)

[garyb]

buffer size will be in the audio app's audio properties page.

you can also set the stm2448's bus outs by clicking and dragging on the number(default is "off") in the cut-out window at the top of the channel, in the middle, under the meters, on the main mixer panel. phew!

son, you have a lot of study to do... the manual found by pressing the "?" on the livebar is really good for knowing what each knob does. take a look at the stm2448/4896 section if you haven't already. just look at the pictures.

one thing that may help the crackles is to shut off all the windows themes that look so cool. right click my computer/properties/advanced/performance push the "settings" button and choose "adjust for best performance". windows eats a lot of resources.

you can press cntrl/alt/del and open the task manager. choose the "processes" tab. find the entry for the program in question (sound forge or wavelab) right click and choose set priority. set it to "above normal'. WARNING!!! fooloing around with this can make your computer lock up or crash. avoid extreme settings and don't mess with anything that you aren't sure about(it's safe to look)!

while you're tracking, just do a rough mix and monitor a single stereo output. when everything is recorded and edited, you are ready to mix. enable enough outputs in cubase that you can send every track or bus, except for things that you may want to subgroup, like a drumset or vocal chorus or horn section, out it's own output. don't forget that stereo outputs are mono if an instrument is assigned to only one side. connect the asio source(Cubase output jacks) to the mixer however. you can stuill use Cubase track autiomation or busses or vsts if you want. since Scope connects the exact same way you would connect a roomfull of gear and you can get in and out of the computer in realtime, you now have a real studio that can interface with anything, in any way that you would like. you can use whatever Scope stuff you want, native stuff you want or whatever hardware you want. you can't really outgrow this, even if you never upgrade your computer.

[garyb]

one other thing.
in a real studio, the mixing desk controls sound several different places.

there is the room where the musicians are, the STUDIO
the room where the engineer and producer are the CONTROL ROOM
the output for the stereo master recorder the MIX OUT
the headphone system MONITOR OUT

the studio and control room have a switching system. they can isten to MIX, MONITOR and EXTERNAL SOURCES like a cd player, the radio, whatever.

you should use the control room out to your monitors. there is a cntrol room volume control, so that you can raise or lower the listening level without affecting the mix level.

[Gary_101]

well, I'm learning bit by bit.

the routing of my analog "in's" thru the big mixer to buses, then out to the sequencer will be a good bit of studio wisdom to incorporate. "keeping it simple" is a challenge with all these crazy wires.

anyway: the issue of crackling when mixing down is still there. In Wavelab, I experimented with all kinds of buffer settings, no effect. Audacity doesn't have that ability.

However there is an interesting little audio app, which I'd forgotten I'd downloaded a trial of, "REAPER". I tried mixing down to REAPER, and oddly enough, it recorded without a single BLIP. This app interests me quite a bit. I like it's GUI.

I'll keep experimenting. Seems like SONAR offers a trimmed down package, a "light" version of X1 ("LE") normally an OEM kind of thing, but it looks like I can still buy it for $59 from their site. This might be fine with XP on a 32 bit system. All the new Calkwalk stuff seems to demand dual or quad core CPU's.

BEHIND all this cracking and popping stuff - I just got a "eureka" - may be this fact: last time I did a clean wipe and re-install of XP, I did not do the alternate install for the HAL (by pressing F5 or 6 or whatever); I think I installed the ASPI thing later on whatever that means. I'm not terribly geek-ish, and unfortunately all these crossed wires and technical finessing take a lot of energy and time, which I would rather spend on creating music. Fortunately, for most of my other work, I switched to the Mac world, and I hate to leave my glorious new iMac to go back to the XP world. but, ah ... music calls.

[garyb]

changing the HAL!!!??? why?

ASPI is obsolete.

[Gary_101]

well, you know that instruction that we got, when you're installing XP, when it asks for something like SCSI drivers or asks you to press F6, you press "F7" instead (i may have that wrong), but you're asked to install a "standard PC HAL" ... whatever. I did it the first time for my P1 and the second time for P1 + P2. I forgot the last time. I feel so guilty. But they also asked you to download the latest ASPI from whatever site. the significance escapes me.

time for a beer, no too early. caffiene. and sunlight.

gary

[garyb]

yes, that's some old stuff.
not applicable with any current software or hardware.

this is the only setup info you need. parts may not apply to your system:
http://forums.planetz.com/viewtopic.php?f=3&t=31345

[dante]

I think the only reason you might use ASPI is to load AKAI sample disks onto your harddrive for use by the Scope STS samplers.

[Gary_101]

sorry for the delay guys; GaryB, the post you mention is for Scope 5.1 on 64 bit systems. I've got 4.5 on 32 bit XP PRO.

I believe the HAL option for XP install is still needed for this type of setup.

Gary

[garyb]

nope.

that post was xp and win7. yes, there is a lot of win7 stuff there.

do you think i've installed more win7 systems than xp systems?

actually, early on with xp, that was a way to get a stable system, but it hasn't been needed since about service pack2. it was really a win98 and dual processor thing.

[Gary_101]

cool! thanks!
saves me a sh*tload of trouble!

gary