It's still cheap, a HDSP9652 would cost about 2k new. But the audiophile 2496 uses digital I/O via coaxial in S/PDIF format. And i am thinking of getting a 2nd hand Fostex VC-8 which has TOSLINK I/O in ADAT format i think....

Which brings me back to my question. Is there some sort of optical/electronic signal converter? Or do i have to get a AD converter with coaxial I/O. If so, any recommendations?

getting a pair of MOTU 896HD for our year end shopping spree.. wee.. christmas came early

will post something about it once i get installed into our system..

Hi, just got a new guitar, so was in the mood for some home recording.

2 guitars, 1 amp, cheapo mic, TOSHIBA notebook. Take a listen, and tell me what you think (of the recording... not my playing, coz the playing is really sloppy)

Disclaimer:
- guitar is slightly out of tune
- crappy amp cone is clipping
- listen at your own risk

http://download.yousendit.com/1A84193A17630D8E

clipping mean u gotta do it again.
the bongos soudn weird to me.

anyway, recording and mixing is different thing.
Recorded nice sound but bad mixing can spoilt the whole thing.
no offend

This post has been edited by limengz: Nov 19 2006, 04:25 AM

recording and mixing go together. Good recording and bad mixing gives the same result as bad recording and good mixing. Bad recording and bad mixing is well....bad (i am not referring to your recording alfee29 because i didn't listen to it. It's a general statement from my big mouth)

The good thing is, good recording and bad mixing can be fixed as long as you have the originals still there. But everybody has to start somewhere, and there's no way the first recording will sound good. My recordings and mixes still suck....really bad.

bassix is right there... lay people usually expects the soundman to perform miracles with bad sound source. It's just a fact of life every soundman has to face. Just finished a small recording project, the instruments sound pretty good but the vocals.. (let's just say if it's possible to delete them, i would.. lol) My point is, try to get it right during the recording stage, or you're gonna have alot of "fun" when it comes to mastering.

limengz: yeah, if I were to put out a cd, definately have to redo. the bongos are from a leafdrum software. i'm a guitarist, new to mixing, learning from u guys!

raist86: is there software out there to "fix" bad vocals after they were recorded? ie, pitch correction or timing correction? i think we can do it manually, but the process is tedious. Correct me if i'm wrong...

honestly, i would rather redo a recording than to use software to correct it. It's possible but tedious and there are some errors that you just can't correct (eg voice projection, mis-pronunciation, lousy eq)

Oh yea, vocal correction can only be done if you did the recording in multi-tracks. But due to hardware constraint, was only able to record the stereo output from my mixer, which is why i say pre-recording preparation is very important.


PS: a little off topic here, but it's regarding drum miking. Found a way to capture our drum kit with just 2 B-5 condensers and a SM 57 with the help of 2 drum shields. what's surprising is that even without miking the kick drum, i still pick it up on the condensers and it sounds pretty tight.

This post has been edited by raist86: Nov 19 2006, 08:12 PM

anyone here know how to reduce latency? i seem to get latency when triggering samples

what setup and samples are you referring to? Example? I just got off a big discussion about latency yesterday. I was quite convinced that a latency of 0.5 ms wouldn't be a problem in a live situation and the salesman was almost killing himself trying to explain to me that latency is a sin.

i) And to lari topic abit, i just stumbled upon a pretty dumb thing today. I now know how to get my signal in to my com, but how does it come out of the com out to my main speakers again?

ii) And supposing my soundcard has more than just 2 (stereo) outputs, can i assign different channels to different outputs (like what i do with the subgroups on a normal mixer) with cubase or whatever software?

iii) Since we are talking about latency, anybody has experiences with really noticable latency problems when using a main mix (from Cubase or whatever software) as a monitoring mix during live performances?

to answer your questions

1) erm, use the stereo outs from your soundcard and line it into the mixer?

2) you can, but you need a Y cable... but it depends on what type of soundcard you have. I had nightmare with audigy 4 pro be4 when cubase will only let me use 1 channel (mono) out. @.@

3) IMO, anything below 5ms is good enough for live recording, especially if you are using a mixer be4 directing the signal to the pc. Heck, you can even use the mixer headphones to do your monitoring if your sound room is in some obscure places.

Thanks raist86, and i have more questions


1&2) - try my best to make my question clear:

Ermm what about ad/da converters, lets say i have 8 inputs (ad) and 8 outputs (da). My 8 signals go into my com via toslink. And then in cubase or whatever software i do a mix. Can i send this mix back to my ad/da converter via ADAT (since it is only 2 channels/stereo) and then say: ok assign stereo mix to outputs 1/2 for main hall speakers, and assign stereo mix to outputs 3/4 for monitoring purposes, and assign stereo mix, for whatever reason, to outputs 5/6 for external...recording or whatever.... Or is my ad/da converter limited, meaning, the mic that goes in to input 1 will also come out of output 1 (which doesn't make any sense to me why someone would build it like that)?

3) Stupid salesman....

EDIT: Soundblaster/Creative soundcards are useless for decent recordings

This post has been edited by Bassix: Nov 20 2006, 04:36 PM

using reason 3.0. when i use reason on its own the latency is very small(but good enough for me to jam with tracks). however when running adobe audition and rewiring(using rewire technology) it to reason, the latency is killing me(not to mention recording is messed up due to the significant lag)

i) You have to enable monitoring. This can be done through your audio interface's mixing software or the recording software. And something which applies to all your questions - IF you decide to monitor using cubase's (or other) monitoring software, there are two options, to use direct monitoring (which will bypass all inserts/eq/effects, only allowing you to change the panning and volume) or to monitor through cubase's mixer (allowing you to hear all your effects). If you use direct monitoring, your latency will be what your interface's latency is - but if you use cubase monitoring, it will add its own latency. This setting for cubase is in Devices > Device Setup > VST Audiobay - "Direct Monitoring" checkbox (I think). Oh and on your track, click the 'monitor' button.

ii) Yes. For cubase, make sure you have made your outputs visible in Devices > Device Setup > VST Audiobay > VST outputs . Enable your outputs there. (This depends on your interface, of course) Then after that, you can go to Devices > VST outputs > group/FX, and then add as many busses and their outputs. Then on your tracks you want to group together, on the strip assign their output to whichever bus you want. Of course, different busses can be routed to various places.

(All this was assuming you have your software set up on ASIO - if you're using windows' drivers, you're naturally limited to 2 inputs)

iii) Latency will always be a problem with monitoring from Cubase (unless you have it set on direct monitoring) - simply because it takes time for the effects to process their input signals, etc. You can try to reduce the latency of your audio interface - by accessing its control panel (or devices > device setup > VST audiobay > select your interface > control panel) - but if you're not in direct monitoring, you will naturally have some latency. Also, again I am assuming, but make sure you're on ASIO!

I'd say that any latencies < 8ms would be okay. My interface has no-latency monitoring so that's not an issue to me when mixing with my interface real-time, but other than that, I have my latency set to 5ms.

Thanks hoongern. I am on ASIO of course. Still unclear on whether i can pair the 8 analogue outputs on my ad/da converter to 4 stereo sets and send the main mix on line level to different amplifiers.

no experience with adobe audition,

Yes, you can - depending on what interface you're using. You set different outputs from cubase itself. (Configured through the device setup > VST outputs) and VST connections.

It's hard to give any really specific instructions without any knowledge of your interface / setup

i have no money and therefore no setup

I am just wondering if it is possible. Before i sell all my bass gear off and get a system that is totally useless.

Assuming i have this setup:

RME HDSP 9652
Fostex VC-8 (it's cheap...that's why )
Cubase 4.0

My 8 input signals come in via the vc 8 onto my system. I know that if i am working with an analogue mixer, i can send these 8 input signals out again through my 8 outputs as analogue signals and then do a mixdown with my analogue mixer for my live mix.

But is it possible that if i am working with a digital mixer or controller, instead of sending a single signal to my outputs, i do a mixdown in cubase and send the mix as stereo through outputs 1 and 2? Do my 8 outputs on the vc8 appear as 8 single line outs on my VST connections list?. Or have I understood the Cubase concept wrongly? I mean after all, i am sending via ADAT optical some sort of data back to my vc8 right?

I have not used the RME before but I'm sure it can do it. But, just for fun, I just created an example below (this is using my E-MU). In this case I set up 8 asio outputs which are routed to the 8 ADAT outputs, as well as 8 ADAT inputs which were routed to adat inputs 1-8.



That is what your VST output list (above) *may* look like. Then (below) you can set up the output busses (I created 4 stereo busses which go to ADAT 1-8)



Then (below) although I have already set up busses, I can also create groups which, well, group certain tracks outputs together and send them to various output busses (defined earlier)



For the individual tracks then I can use different output busses / group outputs (below).



You can see how the group tracks appear in cubase, as shown below



I am currently using cubase 3, so it may differ from other versions of cubase, but the basic ideas should remain the same.

The only thing with mixing live in cubase, again, is that you're going to get latency. It may affect the performance. Perhaps I'm not entirely sure of your question!

Even mixing with my zero-latency E-MU... is a lie. In the digital domain, I experience 12 samples of latency (almost 0.3 milliseconds).. and since you're going to be dealing with A/D and D/A, that's going to add latency. Most chips add around 32 samples of latency. When I mix live (with this zero-latency mixer) my total input+output latency is exactly 222 samples / 5ms (not sure why, though)!!

This post has been edited by hoongern: Nov 21 2006, 03:39 AM

nah you answered my question perfectly. Thanks for the explanations and pics.

Although... i can't really picture or imagine how badly 5ms of latency would affect a live situation. Guess i have to try it out for myself and see if i can live with it. But everything has latency. Even an analogue mixer has a certain latency. My mics have latency...etc etc... Zero-latency really doesn't exist. So these sound people came up with a new term called "near-zero-latency mixing". Like the MOTU 2408 or whatever via firewire to a PCI card where mixing is done in the card itself with some software called DSP.

My next question would be, can i still use Cubase to mix my channels, if the signal is already mixed with this DSP program? I'm not sure what this DSP is, but i think it is the interface for the PCI card that my A/D converter is connected to.

do u guys hav any comments bout my song,plss..

This Is The Songs