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Junior Varsity
How to use VST Plugins live with your X32!?!
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<blockquote data-quote="Rasmus Rosenberg" data-source="post: 74551" data-attributes="member: 886"><p>Re: How to use VST Plugins live with your X32!?!</p><p></p><p>Hey Christian, </p><p>I have done quite a few things like what you describe, all though not on the X32 yet, so sorry for not beeing able to be more specific. </p><p>I find that I want to be in a latency "window" of about 3-4ms max. That kind of gives around a 1,3 mtr of "physical delay". I like to think about it in keeping it in like the distance from the Kick or snare to OH. I find that gives a little more freedom, like processing the kick/snare but not the OH, so you "mess up" the timing but now they are at least "aligned" to the OH. The distance from the Lead vocal to the DRkit or Backline(line), could also define the max.</p><p> </p><p>What I have found is that unless I can run a sample rate at 96hkz the latency will be to great for it to work in most instances (like 90pct of my gigs, from speaker on a stick to Arena). So if only 48kzh able? (don't know the X32) I would stay with just reverb/delays. All though there are plenty instances where a 9ms lag but better processed is an improvement, but off cause also a lot of time where its makes it worse. What im saying is just, be prepared to have a backup plan. </p><p></p><p>Buffer size at 64 is good, don't go lower. </p><p></p><p>Besides sample rate, the host you decide to run your plugins on will have a great effect on latency too. I have measured up to 4 ms difference in running the same plug in, between Waves multirack, Logic, Abelton live, Protools, Garageband, Audio desk, Cuebase etc. </p><p></p><p>There are also a huge difference in latency from plugin to plugin, its worth searching out what gets the job done with the least latency, there are more than enough plugins to choose from out there. </p><p> </p><p>I prefere to run the plugins on groups not directly on the channels, IMO it makes it easier to bypass, I just have to keep track on how to bypass the insert on the group on different consoles. Also be aware of gain structure, its very easy to get confused. I find it very important to setup so I can bypass the different plugins with out dramatic level loss or gain, makes it much easier to compare and "fail safe". Its easier said than done, in the studio 12db of compression or makeup gain is just done, in Live we have headroom and feedback issues that normally don't allow that freedom. So some of the meters and knobs respond a bit quite dramatic.</p><p></p><p>Unless Im touring with a band an have plenty of time to communicate with them, I will not send the processed channels to the monitors. I want to be free a FOH to experiment, and not having the xtra stress of latency issues on stage. </p><p> </p><p>As for mixing processed inputs with straight through, don't do it in the console. Either find a host that has Latency compensation and can split the channel in there and make your processed vs non processed mix in there. Or Split the channel in the Plug in host and use a delay plugin to compensate the best you can. If you want to do parallel compression find some plugins that can do that (mix the compressed signal with the dry), again so you don't have to double buss. IMO its much easier to keep all the processing at the Plug in Host, not on the console , in a stressed situation you could quickly forget the 9ms on the console channel, and send it somewhere wrong.</p><p></p><p>I strongly recommend that you setup,test and measure you setup before gig. Its can be really time consuming to trouble shoot at gig. And the more you know about your setup the more flexible you can be. </p><p>Have fun and keep experimenting!</p><p>/R</p></blockquote><p></p>
[QUOTE="Rasmus Rosenberg, post: 74551, member: 886"] Re: How to use VST Plugins live with your X32!?! Hey Christian, I have done quite a few things like what you describe, all though not on the X32 yet, so sorry for not beeing able to be more specific. I find that I want to be in a latency "window" of about 3-4ms max. That kind of gives around a 1,3 mtr of "physical delay". I like to think about it in keeping it in like the distance from the Kick or snare to OH. I find that gives a little more freedom, like processing the kick/snare but not the OH, so you "mess up" the timing but now they are at least "aligned" to the OH. The distance from the Lead vocal to the DRkit or Backline(line), could also define the max. What I have found is that unless I can run a sample rate at 96hkz the latency will be to great for it to work in most instances (like 90pct of my gigs, from speaker on a stick to Arena). So if only 48kzh able? (don't know the X32) I would stay with just reverb/delays. All though there are plenty instances where a 9ms lag but better processed is an improvement, but off cause also a lot of time where its makes it worse. What im saying is just, be prepared to have a backup plan. Buffer size at 64 is good, don't go lower. Besides sample rate, the host you decide to run your plugins on will have a great effect on latency too. I have measured up to 4 ms difference in running the same plug in, between Waves multirack, Logic, Abelton live, Protools, Garageband, Audio desk, Cuebase etc. There are also a huge difference in latency from plugin to plugin, its worth searching out what gets the job done with the least latency, there are more than enough plugins to choose from out there. I prefere to run the plugins on groups not directly on the channels, IMO it makes it easier to bypass, I just have to keep track on how to bypass the insert on the group on different consoles. Also be aware of gain structure, its very easy to get confused. I find it very important to setup so I can bypass the different plugins with out dramatic level loss or gain, makes it much easier to compare and "fail safe". Its easier said than done, in the studio 12db of compression or makeup gain is just done, in Live we have headroom and feedback issues that normally don't allow that freedom. So some of the meters and knobs respond a bit quite dramatic. Unless Im touring with a band an have plenty of time to communicate with them, I will not send the processed channels to the monitors. I want to be free a FOH to experiment, and not having the xtra stress of latency issues on stage. As for mixing processed inputs with straight through, don't do it in the console. Either find a host that has Latency compensation and can split the channel in there and make your processed vs non processed mix in there. Or Split the channel in the Plug in host and use a delay plugin to compensate the best you can. If you want to do parallel compression find some plugins that can do that (mix the compressed signal with the dry), again so you don't have to double buss. IMO its much easier to keep all the processing at the Plug in Host, not on the console , in a stressed situation you could quickly forget the 9ms on the console channel, and send it somewhere wrong. I strongly recommend that you setup,test and measure you setup before gig. Its can be really time consuming to trouble shoot at gig. And the more you know about your setup the more flexible you can be. Have fun and keep experimenting! /R [/QUOTE]
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