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msmcleod

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Everything posted by msmcleod

  1. It already does this (unless you exclude the Tempo Track from the arranger). But the OP wanted the same section to be played at different tempos... so you'd need to "roll out" the arrangement first.
  2. Having fall back backing tracks should you lose your voice half way through a show, or even a tour seems a sensible thing to do. No need to cancel the gigs, and 99% of the audience wouldn't notice in any case. I suspect they'd prefer this to having the show cancelled. Far better than what they used to do in the 50's, which was to stick some guy who "looked the part" up front miming, while a completely different singer was hiding behind a curtain.
  3. When creating an arrangement, all you're doing is marking sections of your existing project. You can then "Preview" the arrangement, which essentially moves the Now Time around, or you can commit the arrangement. Committing an arrangement is essentially big cut & paste operation on steroids, either replacing your existing project with the arranged song, or inserting your arranged song at a different point in the timeline. For what you want to do, I'd advise the following: 1. Commit your arrangement to a point after the end of your song. 2. Set your Project Start Marker to where your arranged song starts 3. Use the Tempo Track to alter the tempo as required for the part of your project now containing the arranged song Once you're happy with your arrangement you can then use Ripple Edit to delete everything before the Project Start Marker.
  4. Just FYI - the Mackie Control is not compatible with ACT, and instead uses a different method for controlling plugins - if you look at the Control Surface page in preferences, you'll see that the ACT checkbox is greyed out for Mackie Control: Plugin parameter mapping on the Mackie is defined in MackieControl.ini ( usually in C:\Program Files\Cakewalk\Shared Surfaces ). If no mapping is defined for a plugin, the parameters are just presented in parameter order. You can edit MackieControl.ini to define your mapping, but it's time consuming, so I wrote the C4 Mapper utility to edit MackieControl.ini for you: Although it's called C4 Mapper, it works with the main Mackie Control too - you just need to set "Number of VPots" to 8, and restrict your mapping to the top row of pots. Note: C4 Mapper uses the %APPDATA%\Cakewalk\ACT Data\genericpluginparams.xml file to determine what plugins/parameters are available. This file is only populated AFTER you've added a control surface in preferences, and is updated when you exit Sonar/CbB. So if you're not seeing a plugin in the list: Open Sonar, add that plugin to an audio track, then quit Sonar. If you want C4 Mapper to be added to Sonar/CbB's Utilities menu, just run this .reg file: c4mapper.reg Windows Registry Editor Version 5.00 [HKEY_LOCAL_MACHINE\SOFTWARE\Cakewalk Music Software\Tools Menu\C4 Mapper] "Type"="Generic" "ExePath"="C:\\Program Files\\Cakewalk\\Shared Surfaces\\C4Mapper.exe" "MenuText"="C4 Mapper"
  5. You're correct. My bad. I haven't recorded drums in years, so I'd forgotten about the change they made, and I wouldn't be surprised if my kvetching had something to do with their taking it up. My memory ain't what it used to be.... Actually, it was something that had been bugging me for ages, but I think your input gave me the motivation to do something about it. I'm not hugely into comping, but I do stick things on loop record when recording guitar solos and drum parts (I get by on them both, but I'm no guitarist or drummer). Typically I'd just keep loop recording 10 - 20 takes until I get it right. It's impossible to stop recording dead-on the start or end of the loop, so you end up stopping half a measure or so into the loop after the final take. With the previous behaviour this always ended up causing an unwanted split, which was a pain to get rid of. It's even more of a pain when multi-tracking drums with grouped clips. Having identical splits across all takes is essential for comping to work correctly, so the sensible thing to my mind was to simply extend the last take to avoid the unwanted split.
  6. I'm always on a budget (it's in the genes)... but for FWIW I've got: Scarlett 18i20 - on its own you get 8 analog inputs and 8 analog outputs; with something like a Behringer ADA8200, you'll get 16 analog inputs and 16 analog outputs. RME Digiface USB - this has 4 ADAT inputs and 4 ADAT outputs, giving 32 inputs and 32 outputs (plus an additional 2 outputs for your main stereo mix). As ADAT is the ONLY input format, you'll need at least one ADAT I/O interface (e.g. Behringer ADA8200, Focusrite Octopre etc). The advantage of the DigiFace is that you can upgrade your converters/mic-pre's as needed. At present, I'm using my 18i20 as one bank of 8 inputs, and 3 x Fostex VC8's. Two of them (16 inputs) come from the direct outs from my A&H MixWizard WZ3, and the other is switchable between a bunch of pre's... JoeMeek MQ3/TFPro P3, GA Pre 73, and an Alice 828. I've got another Digiface connected to the 32 direct outs from an A&H GL2400-32 via 4 x Fostex VC-8's. I'd like better converters than the VC-8's, but to be honest, they sound great as they are.
  7. @sjoens - you may need to disable USB sleep in the Windows Device Manager. I tend to check in two places: - The Windows Power Plan - The Device Manager for that particular device. Also, some motherboards allow you to disable it in the BIOS, which can save you having to change it elsewhere.
  8. Although RPN #0 #0 is the standard for Pitch Bend sensitivity, there's no standard as to what the ranges actually mean. Some synths have a maximum of two semi-tones for pitch bend range, so the RPN values will be totally dependent on what the synth supports. When using this RPN, the 14-bit range -8912 to 8911, or 0 to 16383 will be for the maximum range the synth supports. So if the synth support 2 semi-tones, either 8911 or 16833 will be 2 semi-tones; if the synth supports and octave, then 8911 or 16383 will mean an octave. You'll need to refer to the synth's documentation to be sure.
  9. You can always try the Yamaha S-YXG50 - it's 32 bit though, and JBridge is recommended when using it: Yamaha S-YXG50 Portable VSTi v1.0.0
  10. The 64-bit Double Precision Engine has nothing to do with the 64-bit architecture. It's to do with the precision of audio data during Sonar's internal mixing. With this unchecked, all audio read from your tracks (or live input) is converted to 32 bit floating point inside Sonar's audio engine. Once all plugin processing/mixing has been done, it's then is converted back to 16 bit or 24 bit (depending on your audio interface settings) when sending to the hardware outputs. With it checked, it uses 64 bit floating point instead. Some plugins however do not work properly with 64 bit audio. Fixed point audio (e.g. 16 bit or 24 bit) has an effective ceiling of 0dB. Floating point however, can go to 100's of dB without ever clipping. 32 bit float has an effective dynamic range equal to 24bit audio, but unlike 24 bit, it maintains this at any volume. 64 bit float has a much, much higher dynamic range. In practise, 32 bit float should be more than enough.
  11. Or change custom so it looks like ALL. You can add as many new ones as you like, so you'd not be losing anything.
  12. I think it's just badly translated - the plugin is clearly disabled in the screenshot.
  13. @MozartMan - Sonar Free Tier will receive updates at the same frequency as the Premium version - it's exactly the same application. New features may not be available in the Free version, but all of the bug fixes will. That's not to say that the free version won't receive new features - just not all of them.
  14. Is this on every project load, or only on the first time you start Sonar?
  15. There are usually options in these VSTi's to control how samples are loaded. Most have the ability to limit the memory they use for samples, and also how they load or stream samples - e.g. do they load the libraries up front, or load (or over just stream) the samples they need on demand. Certainly in Kontakt, if you set it to load on demand, and purge all samples before saving your project, the project will open extremely quickly. If you set it to load all libraries, it'll take much longer to load the project. The downside of load-on-demand, is you may have to play your project all the way through so it picks up the samples it needs, and you can expect glitches during this first playback if the libraries are large. Ensuring your sample library is on an SSD makes a huge difference. Loading a large patch in Omnisphere went from 45 seconds to 2-3 seconds when I moved from HD to SSD (although most patches load in less than a second). You can tweak your memory limit/preview size/streaming settings to optimise this. With the exception of streaming, there's no getting around the fact that those samples need to be loaded however, and they're going to take time to load, so really it's up to you when you want that to be. If you're going to use sample based instruments (and especially if you're streaming), you need plenty of memory and a fast SSD to store the samples to get optimum performance. For streaming, you'll need a faster PC too. Some even go the extra mile and use Vienna Sound Library with their samples on a separate PC dedicated to hosting the samples.
  16. @tdehan - please save your project as a .cwz file, and send a link to it to @Jonathan Sasor.
  17. Delay compensation is still in place when you bypass a plugin, otherwise you'd get glitches every time you toggle bypass/enable.
  18. That is what dragging to the time ruler does - it uses Melodyne to extract the tempo from the audio. This is why you need Melodyne installed for this functionality to work.
  19. Remove the RPN events, and it'll work with SI Piano. With any external MIDI file, you should check the MIDI events and their values are actually supported by the synth you're using. Also, the velocity of the notes is very quiet.
  20. Are you sure you're using Ripple Edit ALL ? It sounds like you're using Ripple Edit Selection, with "When clips are selected, select track automation too" setting turned off.
  21. You need to uncheck both of these in preferences, and do another VST scan:
  22. I quite like the Cerberus Bass Amp from Kuassa.
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