rsinger

To do this inside CbB look up groove clip looping. You can turn audio into a loop. You don't need a sampler to this inside CbB.

A nice feature is that you can bypass the preamp and or power amp and it has a power soak. I have an old VG-99. It has lousy speaker emulation, you had to eq speaker response to get a decent sound. Putting the VG-99 through the Suhr PT-100 with the preamp disabled is nice. Unfortunately it doesn't look like you can load IRs. I think this will be good doing the same thing with an old MP-1 preamp as well.

According to the web the juno gi has a built in IF. If you are using that then you need to select it's driver inside CbB. You may need to configure the IF - roland/boss IFs differ. Speakers/headphones will be plugged into the juno and you need to setup an audio track for audio from the juno (track input) output to master and select Input Echo.

The first thing to do is open preferences and see if the Tascam US appears in the audio devices. If so that's not the problem. Then make sure that audio tracks are routed to a bus. Then check that buses are routed to master and that the master is routed to the Tascam.

Ooops, I haven't done it with USB. Once you have the arduino programmed and you plug it into the computer does it show up as a midi device?

If the motion depth is set to 0 in fxReverb you should be able to set the sonitus reverb to match, but you'll probably have to do some tweaking. If the motion depth is greater than 0 then you need to create a send bus and put the reverb on that and add modulation before or after the reverb. You can use sonitus modulator and try ensemble or string phaser mode on the modulator. HTH.

I have both an M4 and midi express XT and they both work fine. The drivers have been solid for me.

I got this message yesterday on my DAW. The last time I used CbB I was using a Boss SY-1000 as an interface. When I opened CbB yesterday the SY-1000 was powered off and a MOTU M4 was powered on. When I opened preferences the M4 was there, but as expected nothing was checked. Checking the appropriate boxes everything was fine. Seems like it could be a case of switching devices as well - I use both the SY-1000 and M4 in ASIO mode.

I also use it to assist my ear when copying synth sounds from one synth to another or guitar amps and so on.

Easy to test. Create a new project and use PRV to create a click track and then record vocals on another track. Let us know what happens.

I think that depends on the version, I think live 11 dropped support for rewire.

I guess you don't play Fender Rhodes? I usually think of them as paired up with Fender Twin amps. Some guitar fx - wah, fuzz, chorus, phaser, and so on. Depends on what kind of music you play. I think of Hammonds paired with Leslies. I'm not a keyboard player; I'm sure there are others.

It's basically right except for the last bit. Input echo echos the input so you hear the fx, but it doesn't stop latency. Use small IO buffers and low latency fx to try and minimize latency.

I got Omnisphere on sale at musicians friend during one of their 15 or 20% sales. I got it years ago. It seems like I may have paid for an upgrade once and I paid for a moog extension, other than that all the updates have been free. I think it was well worth the money,

The first thing that comes to mind is eq. It's likely you need to eq the guitar for the mix. What sounds good by itself changes in the mix. If it's a lead guitar part - guitarists often boost the mids (around 500 hz). It could also be a problem with dynamics, you can add a compressor to tame the peaks, either before the IF or immediately inside the DAW. If you do it in the DAW then add some make up gain in the compressor and it will raise the level of the track.

If you don't have a guitar preamp you'll want an interface for guitar, something like the axe io. https://www.ikmultimedia.com/products/axeio/ You'll want one of the inputs to have a 1 MOhm impedance. Many IFs will say hi Z input, but they won't be 1 MOhm. An under powered laptop will have more latency. If your laptop is under powered you may want to consider an amp modeler like the nux mg 30. It will give you amp modeling + fx and an audio IF. https://www.nuxefx.com/mg-30.html

I like an amp that has a light crunch, but you can't use reverb with it and you can't do much with delay except a slap back, if that, so if the reverb and delays take a line level signal you can take a DI from an amp and output the pedal to the PA.

It's the Feedback Loop section in these forums.

I think you have it backwards. CbB is audio editing SW. You can import video, but you would compose to the video, not edit the video to the music, you need video editing SW if you want to do that. Video editing SW will let you import still images and manipulate their duration as well as import audio. There may be SW that let's you do both...

It does take time to learn a DAW. I started with cakewalk in 1990 and I'm sure part of the reason I continue to use it is because I'm familiar with it. I also use Ableton Live. HW often comes with lite versions of DAWs and I usually kick the tires, but nothing has compelled me to switch yet. Good luck.

You can start here. Do you have an audio IF or are you using the onboard HW? If you have an audio IF have you installed its driver?

The last company that owned cakewalk prior to Bandlab was Gibson, prior to that Roland owned it, and it was originally developed as a midi sequencer by Greg Hendershott back in the late 80's and he went on to develop it into a full fledged DAW. His company was Twelve Tone Systems. IIRC the original DAW was Cakewalk and later the name was changed to Sonar so that's a product name. I think it should be fine for your purpose. Each DAW has strong and weak points so it's a good idea to try several.

The voltage of an audio signal is also called the amplitude as well as volume and that exists in the digital domain. Note that Millivolts (volts) is the v axis in the picture above. The red line below represents the envelope of the signal displayed. From my experience dsp envelope followers work fine. Just like with HW you have to adjust the gain of the signal and the gain of the envelope follower.

This is not true for Kontakt. In Native Access you can change the content directory to a different drive. They show that if you look at their help. You can also move the Kontakt sample libraries after they are downloaded. I just upgraded from Komplete Select to Komplete and moved the Kontact libraries to another SSD. You just have to repair the links in Native Access after you move the libraries. FWIW.

I don't know the answer to that. I have 2 buffers chosen in "buffers in playback Que". The various asio drivers I have don't seem to have a safe mode option. If you want to get greedy you can use higher sample rates. The higher the sampling rate the more samples there are per second so the same number of samples are smaller units of time. Depending on your current sampling rate you could try going up to 96k hz or 192k hz.