Hi
I currently run logic [PC] 5.5.1 in XTC mode with luna II and A16 ultra + adat expansion board.
recording mainly acoustic instruments but not all.
I have no issues with xtc mode.
I currently have about 4.3 milliseconds measured latency when overdubbing. Although this amount is possible to cause small phase issues it shouldnt be heard as a delay.
Situation is this.
When I record keyboard player using EXS 24 there is noticeable delay making it unusable and I have to monitor direct from keyboard sound source to keep the player in time.
Question 1.
Will using scope projects with a scope sample player have NO latency.
Question 2.
Also can you record through scope mixer using scope vinco compressor and get NO latency.
I can record effects live in logic but it seems there is latency caused by the effect in the chain.
If this was a delay based effect a few MS would be no issue.
Generally im using hardware compressors when recording but at times I dont have enough and want to explore the scope option.
Question 3.
I only have luna II with 3 sharcs...will that support a sampler?
Question4.
I currently monitor live recordings via a 24 channel mixer.
Is it possible to go mixerless and have latency free monitoring?????
Thanks in advance
Maak Bow
<font size=-1>[ This Message was edited by: maakbow on 2004-09-27 17:15 ]</font>
latency questions re move from xtc to scope progects [logic
question 1,2,3 yes,yes,yes
3b: the engine of the STS samplers is rather demanding if the number of voices increases.
So you'd probably need to set the polyphony rather exactly to what's needed.
Imho Gigasampler has a fairly low latency - I dunno the EXS, though.
question 4: if the 16 channels of the A16 are enough and you have micpres in front of the converter you can spare the mixer.
The Luna's (Scope home) analog outs should drive a couple of 16-60 Ohm phones directly.
If the level is too low in a live situation then of course some kind of amp is needed.
your tolerable latency depends much on the situation and type of signal.
For final mixing of signals where the decaying sounds overlap or for the attack phase of layered drum sounds even a few samples may be way over the top.
In those cases the final mix will differ significantly from your monitor signal, which may be a nice or a not so pleasant surprise
cheers, Tom
<font size=-1>[ This Message was edited by: astroman on 2004-09-27 21:29 ]</font>
3b: the engine of the STS samplers is rather demanding if the number of voices increases.
So you'd probably need to set the polyphony rather exactly to what's needed.
Imho Gigasampler has a fairly low latency - I dunno the EXS, though.
question 4: if the 16 channels of the A16 are enough and you have micpres in front of the converter you can spare the mixer.
The Luna's (Scope home) analog outs should drive a couple of 16-60 Ohm phones directly.
If the level is too low in a live situation then of course some kind of amp is needed.
your tolerable latency depends much on the situation and type of signal.
For final mixing of signals where the decaying sounds overlap or for the attack phase of layered drum sounds even a few samples may be way over the top.
In those cases the final mix will differ significantly from your monitor signal, which may be a nice or a not so pleasant surprise
cheers, Tom
<font size=-1>[ This Message was edited by: astroman on 2004-09-27 21:29 ]</font>
I do. I can cook up something if you give me more details (such as what I/O and converters you use, what you are trying/wanting to do exactly, etc.) For monitoring and recording, you can monitor the signal straight in SFP (ie, send it back to the output before it hits ASIO) so that you don't have latency (well, some of it, from the conversion, but usually that's under 2ms if not under 1ms,) and send it thru ASIO to Logic for recording. The version in Logic will be a little delayed, but its pretty trivial to hand-sync it afterwards. I've tried recording in the STS-3000 sampler, but can't say I was convinced.
Also, I'd suggest recording raw signals as much as possible. Altho, again, it's trivial (yes I've spent 8 years in engineering school, what a mess my vocabulary has become!!) to record the raw signal in one channel, and record another channel with all the processing you can dream about (altho a Luna might run out fast, but you can still get stuff going if you don't use a mixer in SFP.) Once the clean signal is recorded, you can process and mutilate it to your heart's content.
Also, I'd suggest recording raw signals as much as possible. Altho, again, it's trivial (yes I've spent 8 years in engineering school, what a mess my vocabulary has become!!) to record the raw signal in one channel, and record another channel with all the processing you can dream about (altho a Luna might run out fast, but you can still get stuff going if you don't use a mixer in SFP.) Once the clean signal is recorded, you can process and mutilate it to your heart's content.
Basicly i record demo's at various levels.
I usually have at least a click track [dr008 or simple klopelfeist]and usually more [exs 24 oes2 etc]before i begin recording.
Mostly recording singer songwriter stuff, keyboard and guitar based. Sometimes I will record a full drum kit 13 or so tracks..other times droo8 or loops.
So the setup needs to allow for all this.
I use A16 ultra converers so have 18 analog inputs available.
Currently Im using a mackie 8 buss [direct outs] as my preamps and monitor/headphone sends. but I am thinking or ditching ther mixer in favour of a bunch of better smaller preamps.
I dont mind using the SFP mixer if necesary.
If mixerless setup/live scope FX setup is succesful I may get the 14 DSP booster board.
I need to be able to create good headphone mixxes.
I only have One out board keyboard and one midi drum pad as controllers though typically the Artists bring in their own keyboards. I only have 1 MIDI input as yet so I do a bit of replugging.
Would like to be able to play samples back without latency during rrecording.
Is that enough info..???
Thanks Heaps
Is that enou
I usually have at least a click track [dr008 or simple klopelfeist]and usually more [exs 24 oes2 etc]before i begin recording.
Mostly recording singer songwriter stuff, keyboard and guitar based. Sometimes I will record a full drum kit 13 or so tracks..other times droo8 or loops.
So the setup needs to allow for all this.
I use A16 ultra converers so have 18 analog inputs available.
Currently Im using a mackie 8 buss [direct outs] as my preamps and monitor/headphone sends. but I am thinking or ditching ther mixer in favour of a bunch of better smaller preamps.
I dont mind using the SFP mixer if necesary.
If mixerless setup/live scope FX setup is succesful I may get the 14 DSP booster board.
I need to be able to create good headphone mixxes.
I only have One out board keyboard and one midi drum pad as controllers though typically the Artists bring in their own keyboards. I only have 1 MIDI input as yet so I do a bit of replugging.
Would like to be able to play samples back without latency during rrecording.
Is that enough info..???
Thanks Heaps
Is that enou
Ok, here's a small idea of how to proceed. First thing to know, latency is incured when the sound goes thru the ASIO drivers. If your signal doesn't go thru ASIO, you won't get latency (actually, you will get a sub-1ms amount of it due to analog/digital and digital/conversion.) So here is the trick:
A few notes: I've used the ZLink modules, replace with ADAT if you use ADAT. Also, this setup is for 8 channels. You can duplicate this to get 16 channels. Also (again), I've used the DynaMixer for this project. I'm not sure if it's still part of the SFP distribution or not. For 16 channels, I suggest you use j9k's mega micro mixer due to its low DSP usage. You should also be able to load the STM1632 mixer so get inserts, pan pots and other gimmicks (has higher DSP use.)
So what happens here, is the signals from the A16 gets mixed and sent straight back to the analog outputs. This will let you hear what is coming from the A16 pretty much instantly (tiny latency incured from the analog/digital/analog conversion like I mentionned up there.) At the same time, the signals get sent to ASIO dest channel 1 to 8, so that they can be recorded in Logic. This is what it looks like on the Logic side:
Notice that the 4 channels with set inputs don't have set outputs. This is to prevent any audio feedback (ie the audio getting sent back to SFP.) With the project I showed, this wouldn't be a problem, as the ASIO sources aren't connected to anything, but say you use a bigger mixer and mix in your Logic tracks while also monitoring sounds, this could be a problem.
So with this trick, the audio gets recorded in Logic, but isn't sent back for you to monitor, since you are already monitoring the signal higher up in the chain. Of course, now that the signal has gone thru ASIO, it'll have a little delay that can be hand-fixed (but you won't be hearing this delay when monitoring.)
I've never used EXS24, so I'm not sure how its inputs and outputs work, but I'm sure you can do a similar trick to record in it. Either telling it to use an unconnected ASIO output, or turning off its output completely.
You can also try using the STS samplers in the SFP environement, which have recording functions. Reading the manuals for these is highly, highly recommended, since the interface can be a serious beeotch at times (no offense meant to any eventual beeotches reading this forum.)
If you use the mega micro mixer, you can probably also load a sampler, and maybe an an effect or two, but nothing really fancy. I'm not sure how much DSP vinco uses, but you should be able to load an instance or two (or more if it's lite on DSP.) You can use a setup like this:
to be able to use vinco as an insert in the DynaMixer, or you can just load the compressors straight in the SFP environnement.
So that should cover it. Feel free to ask any more questions.
A few notes: I've used the ZLink modules, replace with ADAT if you use ADAT. Also, this setup is for 8 channels. You can duplicate this to get 16 channels. Also (again), I've used the DynaMixer for this project. I'm not sure if it's still part of the SFP distribution or not. For 16 channels, I suggest you use j9k's mega micro mixer due to its low DSP usage. You should also be able to load the STM1632 mixer so get inserts, pan pots and other gimmicks (has higher DSP use.)
So what happens here, is the signals from the A16 gets mixed and sent straight back to the analog outputs. This will let you hear what is coming from the A16 pretty much instantly (tiny latency incured from the analog/digital/analog conversion like I mentionned up there.) At the same time, the signals get sent to ASIO dest channel 1 to 8, so that they can be recorded in Logic. This is what it looks like on the Logic side:
Notice that the 4 channels with set inputs don't have set outputs. This is to prevent any audio feedback (ie the audio getting sent back to SFP.) With the project I showed, this wouldn't be a problem, as the ASIO sources aren't connected to anything, but say you use a bigger mixer and mix in your Logic tracks while also monitoring sounds, this could be a problem.
So with this trick, the audio gets recorded in Logic, but isn't sent back for you to monitor, since you are already monitoring the signal higher up in the chain. Of course, now that the signal has gone thru ASIO, it'll have a little delay that can be hand-fixed (but you won't be hearing this delay when monitoring.)
I've never used EXS24, so I'm not sure how its inputs and outputs work, but I'm sure you can do a similar trick to record in it. Either telling it to use an unconnected ASIO output, or turning off its output completely.
You can also try using the STS samplers in the SFP environement, which have recording functions. Reading the manuals for these is highly, highly recommended, since the interface can be a serious beeotch at times (no offense meant to any eventual beeotches reading this forum.)
If you use the mega micro mixer, you can probably also load a sampler, and maybe an an effect or two, but nothing really fancy. I'm not sure how much DSP vinco uses, but you should be able to load an instance or two (or more if it's lite on DSP.) You can use a setup like this:
to be able to use vinco as an insert in the DynaMixer, or you can just load the compressors straight in the SFP environnement.
So that should cover it. Feel free to ask any more questions.