msmcleod

I'm forever putting my back out, and I've got an old shoulder injury I sustained while playing squash which plays up every now and then when putting PA speakers up. My usual go-to is my tens machine, which works wonders. On my shoulder, the pain is gone again for around 6 months... for my back, it takes away enough pain to allow me to move around normally while it sorts itself out over the next couple of days.

Aha - there is a two-click solution.... only just noticed it. Click on the right hand corner of the clip header, and you'll see a popup menu. You can select Render Region FX from there:

I was about to suggest that, but realised if you've got clip gain (or pan) automation, or any clip fx, they'll also get bounced. If you've not got any of those, then yes, it's the same.

FWIW I record everything on MIDI channel 1. The track MIDI channel overrides the MIDI channel before it gets to the synth (whether hardware or software). In any case, if you're exporting as audio to give to an engineer, MIDI doesn't come into it. I wouldn't worry about it.

Up-sampling on Render is when freezing, bouncing or exporting audio, whereas up-sampling on Playback is just when playing the audio in real time. Up-sampling will almost double your CPU usage, so you may want it only for rendering. Your audio interface doesn't really come into it. The way it works is, that audio is read from the track, then up-sampled to double the rate as it's passed through the plugins. It's then down-sampled back to your project sample rate before being sent out to your audio interface. Up-sampling can help reduce additional artefacts introduced at audible frequencies, when a plugin tries to add harmonics above the nyquist frequency (which is half your project sampling frequency). This is generally only applicable to plugins that generate saturation or distortion, and the artefacts although visible on an analyser, aren't (IMO) usually that audible. The down-sampling at the end won't introduce any additional artefacts as this is a simple process. It's worth mentioning though that: - Many modern plugins do internal oversampling, which can negate any benefit of up-sampling. - Not all plugins will work at an up-sampled frequency - some may even crash, especially your project is at 44.1K (88.2K is not a common sample rate). - Many plugins will sound exactly the same after you up-sample them, but will take up twice the CPU usage. So my advice would be, check each plugin first to see if it works, and if it makes an audible difference - if it doesn't, don't bother.

You can also bind a keyboard shortcut to open the synth on the active track:

The Realtek ASIO driver is the problem. It doesn't even pass the most basic ASIO tests using the test application in Steinberg's ASIO development kit. If you're using a Realtek device, use WASAPI Shared if you want to share the device with other apps, or WASAPI Exclusive for low latency.

I guess it all depends on what gear you have, but having gigged with various instruments, I would say from my experience: Easiest: Sax then.. Guitar ( 2 guitars, 2 stands, VG88, Laney Amp ) Keys - ( 2 or 3 keyboards with stands, 8U rack with sound modules, outboard mixer, keyboard amp ) Nowadays my Keys rig is cut-down. I can get away with one stand, a Studio Logic SL880, Korg X5D, SMPro V-Machine, a passive mixer and a keyboard amp.

Glad it worked out for you. If possible, could you capture a crash dump and send it to us with the plugin flushing set to 20? It would be useful to know which plugin is causing the issue.

It's not confusing when you read the paragraph afterwards: This statement is incorrect. As I said in the last paragraph of my original reply: If you set an output channel for the track, it will send all MIDI events out on that MIDI channel, regardless of what channel the MIDI events are. If the track output channel is set to "None", only then will it send MIDI out on the channels they were recorded at.

Plugin flushing sends silence to the plugin for a period of time, or until a quietness threshold is met. When set to 20, it's the equivalent of 20 seconds of silence. This is to ensure that things like reverb tails are cleared before moving on to the next export. Some plugins get upset with the plugin flushing, so setting it to 0 essentially disables it.

The time range is ignored for exporting arrangements. Alternatively, you can export the arrangement directly from the arranger inspector:

I'm not 100% sure what you mean here... do you mean the duplicated track isn't making a sound? Have you selected events in the duplicate dialog? If you want an exact replica, you need to check all the boxes ( apart from Link to Original Clip(s) - unless you want that ).

You can mute previous takes if Auto-Punch is enabled. Actually this is what I quite often do to get around the split behaviour in comp mode. I just set the Auto-Punch in to be where I want to start recording, and the Auto Punch out where I want to stop, or if I don't know, a 100 measures or so after. This was the old way of doing multiple takes before the comp mode was introduced.

@dogufo - so this is how I'd do it: 1. Set the input port for each track to whatever keyboard you've got the MIDI In set to, and set the channel to Ch1 - e.g. fatar MIDI Port 1, channel 1; SY77 Port 2, channel 1. 2. Set the output port of each track to the SY77, and the MIDI channel to whatever sound you want played on the SY77. The MIDI channel of events will always be channel 1, but if the output channel is set on the tracks in CbB, it'll send all events out on that track to the channel you've set. So in essence, ignore the MIDI channel in the event list - they'll always be overridden by the track channel if you've set one.

The easiest way around this is to create an arranger section that covers the area you want to move, then move that arranger section. Arranger section moves will move everything, including bus automation.

FYI - I brought this issue up in meetings, and the general consensus was that rather than splitting it, the default option should be to extend the last take to the loop extent. I did spend some time looking at it, and it wasn't as straightforward as I'd hoped, so I'm not sure when we'll get around to fixing this.

FWIW I did a recent DAW machine build and found that Dimension Pro only authorized after I ran SONAR Platinum and inserted it as an instrument track. I can't remember if I had to run SONAR as administrator or not, but basically the steps were: 1. Opened CbB, inserted Dimension Pro - it wasn't authorised 2. Closed CbB, opened SONAR Platinum and inserted Dimension Pro - it showed as authorized 3. Closed SONAR, and opened CbB and inserted Dimension Pro - it was still authorized.

Just in case you didn't know, you can use any VST in the ProChannel using effects chains: You don't need to map the controls like the example below, but it can be useful if you want easy access to them:

Have you tried this: 1. Setting the Render Buffer size to "Playback Size": ... and if that doesn't help, 2. Disabling Plugin Flushing:

No, however AFAIK the latest Softube update has re-enabled ProChannel support for the plugins that worked as ProChannel modules. IIRC these are the TSAR-1 R Reverb, Passive Equalizer, Active Equalizer, Focusing Equalizer and the FET Compressor. Both Boz Digital Labs and Overloud also did ProChannel modules, but I don't think they're available any more.

Given the problems with the logarithmic behaviour of volume, I'd have thought it easier to just draw an automation curve on one channel, and draw the reverse on another.

A CbB-only solution might be to use track templates, where all the audio tracks for the synths are in a single folder. Another 3rd party solution might be Cantable, but personally I've only used this across machines via ADAT and not on the same machine as CbB.

You need to turn Global snap on... ...and set the snap resolution to the appropriate amount. For example, if you are in 4/4 and you set the snap resolution to 1/4, the bar within the staff view will be split into 4 hit-areas, and your note will be placed at the nearest 1/4 note. Note that snap setting only affects the start time - the duration of your notes is whatever you've selected in the staff view toolbar. The reason it looks right on the screen, is because Cakewalk automatically quantizes the display to whatever you have set as your staff view display resolution: This is to support a sensible representation of parts that are not quantized.