Looks like you might need to double your tape speed or half your sample frequency if the max is 10k samples.Beneficial wrote:Am I right in thinking this plug-in only works when recording at 44100? I tried this last night at 88200 but my delay was over 12,000 samples long and it looks like the max compensation with the plug-in is 10,000 samples. Cool idea though... hoping they release a new version of this.
A cool way to integrate analog tape with your DAW
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Hmm... well I'm using it with my JH-110 running at 15ips capturing at 48k. So that would be analogous to running at 30ips and capturing at 96k. What's the delay that you are trying to put in there? What kind of tape deck do you have?
I haven't tried to use the plugin at a higher sampling rate, but as far as I know it should work fine at 88.2 and 96k.
Can you post a screen shot of the two waves you are trying to measure the delay between? Perhaps you have something funky going on? Just as a sanity check how far apart are the record and playback heads in inches?
thanks,
Brad
I haven't tried to use the plugin at a higher sampling rate, but as far as I know it should work fine at 88.2 and 96k.
Can you post a screen shot of the two waves you are trying to measure the delay between? Perhaps you have something funky going on? Just as a sanity check how far apart are the record and playback heads in inches?
thanks,
Brad
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Just doing some math in my head...
If you are recording at 88.2k then 10000 samples gives you a max capability of 113 ms delay between the two heads. At 30ips that means your record and playback head can be no more than 3.39" apart. So it is feasible that you are exceeding the capability of the plugin. Let me email the Voxengo guy and see if he can make a revision to the plugin.
thanks,
Brad
If you are recording at 88.2k then 10000 samples gives you a max capability of 113 ms delay between the two heads. At 30ips that means your record and playback head can be no more than 3.39" apart. So it is feasible that you are exceeding the capability of the plugin. Let me email the Voxengo guy and see if he can make a revision to the plugin.
thanks,
Brad
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Brad,
First of all, this "trick" of yours? COOLEST. SHIT. EVER.
I mean, this really opens up broad new worlds for me and I'm not sure if everyone here's really embraced the ramifications of this. I think to Toaster's question about how this really is a benefit over just recording to tape can really be answered by saying...
The combination of the Voxengo plugin with the software monitoring scheme is basically a step away from time travel: an amazing way to function in your DAW in a very business-as-usual way and your deck always running in the background makes it pretty much like strapping a plugin across your channel...except it's REAL TAPE. The fact that you can get "that sound" and NOT as an emulation, and yet also continue to do DAW edits like a madman in real-time is just shy of miraculous even though the basic principles are quite rudimentary!
First of all, this "trick" of yours? COOLEST. SHIT. EVER.
I mean, this really opens up broad new worlds for me and I'm not sure if everyone here's really embraced the ramifications of this. I think to Toaster's question about how this really is a benefit over just recording to tape can really be answered by saying...
- ? Yes, many of us don't have the luxury of 24 or even 16 channels of high-quality tape
? Even if we did, the ramifications of being able to run different formulas on different machines simultaneously is serious boner material
? If one is not already really accustomed to the world of non-linear editing, he/sher may not understand just how fast, diverse and powerful a tool that is, nor how seamless punch in/outs become
The combination of the Voxengo plugin with the software monitoring scheme is basically a step away from time travel: an amazing way to function in your DAW in a very business-as-usual way and your deck always running in the background makes it pretty much like strapping a plugin across your channel...except it's REAL TAPE. The fact that you can get "that sound" and NOT as an emulation, and yet also continue to do DAW edits like a madman in real-time is just shy of miraculous even though the basic principles are quite rudimentary!
"All energy flows in accordance with the whims of the great Magnet"
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And furthermore...
I wanted to put this in a seperate post so it wouldn't get lost in the shuffle as it's technically important:
I went to the Voxengo site and was bummed to find (unless I'm missing something) that the plugin is only for PC and only in VST format.
I'm on a Mac using Digital Performer, so I did a little digging and came across an Audio Units take this same idea, called LatencyFixer by Expert Sleepers. It's at...
http://www.expertsleepers.co.uk/
It appears from Brad's tutorial as though the Voxengo plug actually displays the affected waveform as if it's been physically shoved back on the timeline the programmed amount. I think?
In any case LatencyFixer plug doesn't seem to do this for what it's worth: it does it's work in the background but the waveform appears the same as before, which is to say off-register with everything else.
I suppose regardless of which plugin you use, it would be a good idea...custodially-speaking...to select all your "from tape" tracks and actually move them forward the necessary amount after your session is done and then disarm the plugin instances...both to avoid confusion later on and also to cut way down on processor strain.
My 4 cents
I went to the Voxengo site and was bummed to find (unless I'm missing something) that the plugin is only for PC and only in VST format.
I'm on a Mac using Digital Performer, so I did a little digging and came across an Audio Units take this same idea, called LatencyFixer by Expert Sleepers. It's at...
http://www.expertsleepers.co.uk/
It appears from Brad's tutorial as though the Voxengo plug actually displays the affected waveform as if it's been physically shoved back on the timeline the programmed amount. I think?
In any case LatencyFixer plug doesn't seem to do this for what it's worth: it does it's work in the background but the waveform appears the same as before, which is to say off-register with everything else.
I suppose regardless of which plugin you use, it would be a good idea...custodially-speaking...to select all your "from tape" tracks and actually move them forward the necessary amount after your session is done and then disarm the plugin instances...both to avoid confusion later on and also to cut way down on processor strain.
My 4 cents
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Dagnabbit yer still not pickin' up what I be layin' down! (Sighs) It's all about the ability to do non-linear digital style editing to tracks recorded to tape in real-time as they happen. I absolutely DEFY you to be able to locate your client's clams as quickly and easily on tape as I can looking at the waveforms themselves and the markers I create on-screen to track the song passages, and grabbing my playline and dragging it back-and-forth along the timeline...along with the "scrub" feature which admittedly is an emulation of tape technique.toaster3000 wrote:well.... the best way is just to track to tape and then dump it
I mean I get what you're saying about being so fast at what you do with tape (having shitloads of real-world experience working in that medium no doubt) that by the time you find the spot the guitar player is still barely done picking his ass and all that, but STILL...non-linear editing is just a whole 'nother beast.
And also the question of the "cleanliness" of punch ins/outs to tape: I know that on MY machine (Otari MX5050) punch-ins are pretty clean if you know what you're doing, punch-outs are a huge slopfest. Other machines I've heard are the reverse of that. And I'd bet the real NICE machines out there...your Studers and MCIs et-al probably totally rule at both though I've never worked with 'em.
That said once again in DAW-land, punch ins/outs are ALWAYS seamless regardless.
I should also point out that in addition to what I said before about those of us who either don't have 24 tracks of tape OR would prefer this method for the ability to use different formulas/speeds simultaneously, one could also argue that the DAW allows a single human being who is recording himself a whole lot of ease in terms of being able to set punch in/out points ahead of time for hands-free clam fixes when the engineer is also the performing musician. I know my old Tascam 4-track in high school had a lovely footswich for this, but I'm not sure if the big decks can do this? I know my Otari can't. So again, there you'd be using a DAW to record your own stuff solo with ease and yet getting that sweet, sweet, tape.
So yeah I know this is coming off like a list of ways that DAWs are better than tape so believe me when I say that at heart I too am a tape purist and love working with it...I think it really increses your skills as a critical listener, a "custodian", a gatekeeper, and all that. Hell, I even love the SMELL of it! But still, I think of all the ramifications of being able to use this new angle in my daily recording life and I just start to sweat like I'm talking to that hot chick at the bank...
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Reaper has a function for automatic latency compensation for inserts. Called ReaInsert.
Could one just treat the tape as an "insert", and use it that way instead of the Voxengo plugin?
Could one just treat the tape as an "insert", and use it that way instead of the Voxengo plugin?
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Re: And furthermore...
This is not correct. You need to instantiate the plugin on your input channel so that as the signal is recorded to the DAW it gets processed by the plugin and slid back in time. Does that make sense? The Voxengo plugin isn't drawing any waveforms. Your DAW is doing that! If you put the plugin on the audio channel then you will not see the waveform in the correct position. Certainly Logic has input channels that are separate from audio channels?Caldo71 wrote:It appears from Brad's tutorial as though the Voxengo plug actually displays the affected waveform as if it's been physically shoved back on the timeline the programmed amount. I think?
In any case LatencyFixer plug doesn't seem to do this for what it's worth: it does it's work in the background but the waveform appears the same as before, which is to say off-register with everything else.
I suppose regardless of which plugin you use, it would be a good idea...custodially-speaking...to select all your "from tape" tracks and actually move them forward the necessary amount after your session is done and then disarm the plugin instances...both to avoid confusion later on and also to cut way down on processor strain.
My 4 cents
Brad
Last edited by Brad McGowan on Mon Aug 25, 2008 11:56 pm, edited 1 time in total.
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Nope. Because you are not using it on an insert. You are using it on an input channel. See the difference? The idea is to record the tape sound to the DAW and have it draw the waveform in the correct location in time.roygbiv wrote:Reaper has a function for automatic latency compensation for inserts. Called ReaInsert.
Could one just treat the tape as an "insert", and use it that way instead of the Voxengo plugin?
Watch the videos again.
Brad
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Re: And furthermore...
Actually I'm in Digital Performer, and well, I'm not sure if I DO get what you're saying as I think the idioms particular to our DAWs of choice may differ quite a bit.Brad McGowan wrote:This is not correct. You need to instantiate the plugin on your input channel so that as the signal is recorded to the DAW it gets processed by the plugin and slid back in time. Does that make sense? The Voxengo plugin isn't drawing any waveforms. Your DAW is doing that! If you put the plugin on the audio channel then you will not see the waveform in the correct position. Certainly Logic has input channels that are separate from audio channels?
Brad
From what you just said, It SOUNDS like on your DAW (Cubase, right? I will go watch the vids again), there's the opportunity to use plugins to "destructively" process audio AS it comes into the computer, so that by the time that you're looking at the waveform in audio channels, the work is already done and there is no plugin at all strapped across that channel of your on-screen "mixer" per-se. Am I getting it?
I'll quiz my DP genius compadres but i THINK that is just not the case in DP.
Now you CAN MONITOR your incoming audio through the plugins at the expense of computer processing power to fight internal latency (like any other DAW out there). Normally I monitor directly from my hardware instead. But that can't be what you mean: it still wouldn't push that waveform back in the timeline by the time you see it in your tracks list. And I'd assume you wouldn't HEAR anything happening by monitoring through the plugin either, unless it folds time and space.
Sorry if my posts are overly-long...the language of this stuff is tricky and all these DAWs are disparate in their methods! The way I went about it DOES seem to be working, though. But visually it's not as correct as what you have achieved because the waveform is in the wrong place even though it plays back properly and my punchins all land in the right place.
Again, I will ask around with those that know DP inside and out and see about applying plugins to the input channels. If there IS a way to do it, it would be cool to get a second tutorial going apropos to DP.
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I think you are getting it. Right--Cubase and Nuendo allow you to destructively process the audio with plugins as they enter the DAW.
As for monitoring goes, I am not using the DAW software to perform monitoring (which introduces latency). I am using the audio interface hardware (RME Fireface in my case) and its associated DSP-power zero-latency monitoring application (RME TotalMix) to handle all monitoring. I think monitoring through plugins messes with automatic delay compensations ability to keep everything in sync so it would not be recommended.
Let me know what you find out about input channels in Digital Performer. If it doesn't allow you to do this then it's time to upgrade your DAW.
Brad
As for monitoring goes, I am not using the DAW software to perform monitoring (which introduces latency). I am using the audio interface hardware (RME Fireface in my case) and its associated DSP-power zero-latency monitoring application (RME TotalMix) to handle all monitoring. I think monitoring through plugins messes with automatic delay compensations ability to keep everything in sync so it would not be recommended.
Let me know what you find out about input channels in Digital Performer. If it doesn't allow you to do this then it's time to upgrade your DAW.
Brad
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