reginaldStjohn

Yes, that is what the punch in should do. Have you looked at pp 279 in the Reference guide on Punch recording? Or online? https://www.cakewalk.com/Documentation?product=SONAR X3&language=3&help=Recording.26.html

In studio one you can "route" your VSTs in the plugin bin so that you can create splits, merges etc. and then move or add plugins to any spot along the path. That way you can do a parallel effect and control the wet dry mix even if the plugin does't support it naively. It is really just like adding a bus or aux track but all done withing the channel.

Generally speaking, most forumites here don't recommend using ASIO4all. I think you can do most of what you want just by switching devices as you need them.

If you mean hear audio on the track you are recording to then you might have to enable "sound on sound" mode of recording by right clicking on the record button. If you mean other tracks then they should be playing all the time. Just set the playback head a few bars ahead of the punch in point or set a loop region larger then the punch in point

ASIO only lets you have one device active at a time. Using ASIO4All just puts a wrapper around windows driver modes so that ASIO can look like it is using more then one device at a time but it is really just using windows type drivers (WASAPI) underneath. Having both installed (ASIO and ASIO4All) can cause problems. To use your Spark device try un-selecting your other devices and then the Spark should become available. If you need to use multiple devices at the same time then switch your driver mode to WASAPI shared mode. However, be aware that multiple audio devices are not necessary time synced to each other and can have different latencies and timing anomalies.

Bundle files were not meant to be used as a working project format. It was meant to "bundle" up a complete project with the associated audio to share, move or archive. You should use the standard .cwp format for continuous work and saving.

Most audio apps are not written to use GPU resources for audio. They can help with rendering some plugin or UI graphics but currently the graphics card is not a major contributor to DAW performance.

This can depend on the synth into which you feed the midi data. In general if you overlap the notes slightly this will make it sound less staccato. However, synths, pianos, virtual instruments often have a legato articulation where it actually slurs (hammer on, pull off, slide pitch bend etc) the note to mimic a human playing the instrument better. These can be triggered by holding down a particular key (called a key switch), the velocity of the note or other midi controls depending on the chosen instrument.

You can put in a feature request in the feedback loop. However, I do have to ask, is it really that hard to to click one button to disarm it? As an alternative you might be able to use "AutoHotkey" macro to disarm all tracks and then arm the active track all with one key press.

I used a program called "Media Monkey" for a while. There is a free version that did everything I wanted it to do with tags. Wave files and CDs do not have the same ability to add images and tags. That is what the CDDB website was for. To look up an inserted CD or folder of Wave files and let a program display album information.

Could also be effects in the Pro Channel fx item instead of the FX bin

The only thing I can think of is that the project file is corrupted somehow?

I haven't done this across projects because I always adjust them visually and by ear. However, If you open the inspector ('i' key) there should be a start time listed for the clip if you select it. You could select the non-moved clip, note its start time, and then note the time of the moved clip.

Try a different driver mode (MME, WASAPI, ASIO) and see if it shows up. Did you install drivers for it?

Yes, you should be able to do remote desktop from a Surface to a PC. Although, it might be a bit overkill for just starting and stopping the project. Some people just use their wireless keyboards for this. There are also apps like "TouchDaw" that you can run on your phone or tablet to remotely control playback and recording.

So is the project in a bundle file or just a normal project on your hard drive? If it is on your hard drive then the audio files should be still there. I always use per project audio files so all the files are in a folder by where the project file is. If your project is in a bun file an it is corrupted then I don't know how to fix that.

You can select a range and with your cursor towards the top of the clip left-click and drag up and down and it creates a 4 node square to raise and lower that section.

That sounds like a Fishman issue to me. Sounds like they don't allow 2 instances of "TriplePlay" to be opened. Isn't Triple Play a midi controller for guitar? If it is then it seems like you would just set it up like a midi input device and then set the tracks input to receive from the Triple Play. Is that what you are doing? Maybe someone else who actually has the Triple Play can respond

From what I read the unit is "buffers". I don't know which ones how how big they are but it is telling cakewalk to to reserve more data buffers for plug-in processing because certain plugins take the available ones and hold on to them. This seems to only be needed if you are experiencing drop-outs. I don't think it would do anything to help with latency As for your other two questions 1. The description seems pretty clear. Turn it on if you want to bypass the cache of your hard drive. I would only do this if you have some specific reason to do that. 2. Only one of the developer's can answer that. However, I don't think you have to know how it works to use it. Try it and if it works better for you then keep it. The current recommendations seem to be keep it at 2

You need to do some investigation to see why this is happening. Try changing the Driver Model to something other then ASIO (MME, or WASAPI). Run latencymon on your computer to see if anything is interrupting the system and see if there are issues that will prevent realtime streaming of audio. The built in audio devices, (Realtek) are always going to have some quirks with them since they are not designed for low-latency DAW type work.

Could be this?

It looks like you comped in a take that was muted. You can un-mute using the mute tool.

I don't know if there is any "quick" help but getting low latency while recording requires you to set the buffer size on the Scarlett lower. Use the windows control panel to do this. Cakewalk often can't directly change the buffer setting so you have to use the windows or vendor supplied way of changing the buffer settings. Now, the no sound issue is another one. Start a new project and make sure that the audio interface and cakewalk are set to the same sample rate and depth. Then record something or drag in an audio file. Make sure your tracks are routed to the main bus and the main bus routed to your Audio Interface outputs, assuming your audio interface is setup correctly in Cakewalk's settings.

I agree that the work arounds are not that great. I can slip edit (Move the audio within the frame of the clip) in the take lanes fine. However, Moving clips or changing the size of the fades in the take lanes does not work well for me.

Try to post some screen shots of your routing and setup and tell us a bit more about how you inserted the VSTs. I am not familiar with Grace. Is it an instrument or an effect?