John Vere

OK that's a big help, no problem with languages as we are used to figuring that out best we can. It would also be nice to know which model the Edrol is. Edited, see my next post.

Exactly why I leave my tracks alone. If nothing is peaking and the mix is very close to my liking and then all that automated stuff as well as level sensitive effects. Example if you pull down the vocal track and it has a send to the reverb bus you have now changed th e send level as well. It should be proportional but there's never a guarantee of this in my experience. No, better to use sub busses for backing off on a mix. I will only return to tracks to balance a sub mix or deal with EQ issues. Overall there is a zillion ways to get to the same place so one just has to find a way that works for them. Ultimately you need a firm understanding of the signal path because making adjustments in one place has a different result than when you might do it somewhere else. This is especially true of the effects bins, Pro Channel and sends. My method works for me on hundreds of mixes. They are far from perfect but they meet my needs and expectations. I didn’t mention this in my first post as it was already turning into a novel. I also put Span on my sub busses. It’s a light load and at the master stage you can always push your buffers up. Actually I just drag Span and Youlean from bin to bin most times

Need more details of the devices you are using and how you have them connected. Can’t even begin to make a guess. All I know for sure is you are playing a guitar and a drum machine is also playing.

Well that certainly is a very old card and possibly you are not using W10 but Windows 7 A new interface is definitely worth thinking about They all have direct monitoring but they also need a blend controller which the cheaper ones are missing. This is a Blurb I made on the topic- Shopping for an Audio Interface? There is a lot of choices. Make a list of your requirements first.. What kind of connectivity do you require? and how many of each. This is the most important determining factor. Most of us only need a few in/outs. It’s nice to not have to unplug and plug stuff in. It’s false economy to short change yourself on this feature. You end up having to purchase a mixer as example. These are some of the options: XLR, or 1/4", Combi jacks, RCA, ¼” line level, ¼” Instrument level , MIDI, SPDIF, ADAT, MADI How many ins and outs do you think you'll need now and in the future? Are they accessible? Front or rear panel? Are the outputs all ¼” jacks Balanced? Some have RCA jacks in Parallel with main outputs which is nice. Is there a true stereo input pair? Some don’t have a matched set of inputs which sucks for recording stereo devices like hardware synths and drum machines. Are there peak level meters or just a little LED for each input? Do all inputs have a peak indicator? Channel Insert jacks are rare but a must have for those with Hardware compressors. Are there separate controls for Speaker Monitor level and headphone level? Is there a blend control for mixing Input Source with Computer ( DAW) ? A lot of cheaper interfaces are missing this important feature and only have a on/off toggle. They might use Software but having it on the front is defiantly the best option. How many Headphone jacks? A level for each? This is also important if you work with other musicians. This will save you purchasing additional hardware. Are the input pads or line / Instrument toggle switches on the front, back or software controlled? Is it a metal box or cheap plastic? Is it light and portable or large and bulky, Rack mountable? Does it have an on / off switch? Seems like a no brainer but you would be surprised how many don't. Does it use Buss power or a power supply? Buss power can have issues with noise and Phantom power and some need a dedicated USB 3 buss. Look for at least an optional power supply. Does it have DSP effects built in? Does it use a GUI mixer? Having a software (GUI) mixer adds more options. Can it be used as stand alone? Some interfaces are also handy as a small mixer. Does it have a Loop back function. This is a newer feature that is real important if you do screen captures or wish to record any playback from outside your DAW. Like using a stand alone VST. Round Trip Latency (RTL) specs are hard to find. Do you need low RTL for real time processing like Guitar Sims? Low RTL is going to be at a higher price point. A $200 interface will have hidden buffers etc. Zero latency is just marketing hype for monitoring directly from the interface. All interfaces have latency and the amount will depend on your buffer settings. Most under $500 interfaces run about 10- 18 ms RTL at 256 buffers. And most important of all, Does it have top notch ASIO drivers for your OS. What is the word on support from the company? Visit the web site and try creating an Account before you purchase. Pretend you just bought the device and see if the drivers and manuals are easy to get at and are kept up to date. Does it come with free software, Example Focusrite interfaces seem to come with a lot of good stuff. This has added value to the purchase price. Everyone will recommend the interface they have chosen, that doesn't mean it is the right one for you. Make your list starting with the input /output specs. Example you determine you need a 4x4 interface. That just narrowed down the search by a lot! You might find only 10 models. Now compare those to get the most features needed.

In my signature is a link to my tutorials Watch the one on using on board audio. Even though you’re using the ASIO sound card the info is the same. There’s also the one on ASIO which will explain that side of the issue and tips on troubleshooting I noticed that you chose 64 bit process. You don’t need that Also try MME mode for midi driver Also uncheck MS Wavetable you don’t want to use that pos this will also allow midi files to open using the TTS 1 My guess is your master bus is not set to the correct output

First the list in the browser is your audio effects VST’s not the instruments. You selected the wrong tab. And the add instrument dialogue has a drop down menu with the other instruments. You can also add the synth rack view to the browser and click the + sign to add a synth in my signature is a link to my tutorials which most are about using midi

Yes and No. The key word you used is the "SUM" as long as no tracks or busses are clipping this is correct. The master is where you set your ultimate output level. At that point it becomes a personal preference of how you want your mix to sound and most importantly the overall loudness or LUFS. as that is a common standard for pop music on the internet. Peak is only a small part of mastering. In this screen shot you can see how I set up my Master bus and use these 4 tools to figure out visually what is going on. You can't always trust your ears or monitoring system. Looking at this I see I need to adjust my bass as it is pushing the Multi band Low MId band too much. I also see my true peak level is past my -0.8 which is what I set the limiter for. My goal is to only see random action on the compressors. My LUFS is just a little more than my goal of -14 as a result. So I start here and will work my way back first to the sub busses and then to the tracks to fine tune the mix and the balance of frequencies. In other words just using peak level is not going to result in a good mix. Peak is just what you need to keep an eye on track by track, bus by bus to make sure there is no RED showing. The trick with compressors and limiters is learning how to not over use them. You might want that compressed sound for vocals, drums and guitars, but generally that's a bad idea on a whole mix. I can simply turn down the input gain of the Multi band which is the same as turning down my master, but as you see the other frequencies would then become too quiet. The multi band is only 5 bands so that is why I use Span to pin point hot frequencies. You can really see the hump in the low end getting flattened which verifies why the multi band is working hard on that band. My bass is being over compressed and hogging the loudness. I can move Span to the top of the stack or even put a second instance there to investigate further. But these tools to me are indispensable and I've had real good results since I adopted this a year ago. My stack is LP Multi Band ( came with Sonar but there are lots of these around) The BT Brickwall set at -1.0 Also a Sonar freebie but the LoudMax might work. Mike of Creative Sauce videos fame seems to like it a lot. You lean Loudness meter- free to use but has a few features disabled so I paid for it. https://youlean.co/youlean-loudness-meter/ Span this is free. https://www.voxengo.com/product/span/

That won’t work unless the op has very low Round Trip Latency. They will hear a delay which is annoying

Better yet you just need to put a limiter on the master and set it at the desired max level. The Loud Max is free and works well for this. https://plugins4free.com/plugin/2060/

Your audio interface needs to have direct monitoring I don’t think those do. So then you will need a mixer with a headphones jack. I’m surprised a Delta card would even have a driver for W10

Ok if you both have USB mikes you could try that. Try WASAPI mode to show your inputs But you need to realize how much latency there will be on his mike and adjust that track. You won’t be able to sing at the same time either

Are you working in the Piano Roll View? You right click a note to delete or hold right mouse Lasso a group of notes and let go and right click on one of the highlighted notes. You should really take some time to watch tutorials as you asking a very basic question which is fine, but it's best to teach yourself as much as possible first and then ask questions when truly stuck. There's a Tutorial sub forum and I have dozens of midi tutorial as well. See my signature for links. https://sites.google.com/view/cactus-studios/cakewalk-videos

What you could have done to reduce the file size is to just copy the project to make a new version using "save as " with copy audio checked, and cut out all but about 10 seconds where the problem was. If you don't know how to do these it's easy as pie- set the now time 6 seconds before the issue spot, Crtl A to select all. "S" key to split. This by default will leave everything to the left highlighted. Delete. Now move the now time just past the problem spot and hit CTRL A and S again but now highlight all the material to the right buy holding down CRTL while you click on each track and delete. Either that or go into Preferences and change to select right of split as in my screen shot. Of note is this is also where you can adjust how splitting a clips crossfade can be altered. And I like to use the blend old with new so I can slip edit splices to get rid of artifacts which probably was all you really needed to do.

Cakewalk or any DAW will not create a distorted clip unless there is a reason. I have no clue why your had an issue but as said most of us when we encounter this assume it is something we have created in audio of the DAW and it needs to be eliminated. As @bdickens said you proved user error, not software, when you eliminated the track that was causing the clipping. End of story.

If you had Sonar and the Blue tube stuff the BT Brickwall is the only limiter I've used for a long time now and it works solid. If I set it to -0.8 all measuring tools will report that that is exactly what I got, The Loudmax is also a great limiter for free, I use that on the output of my Movies in Vegas.

OK Cubase- this means you also probably ended up with the Steinberg Generic ASIO driver as your Recording Latency adjustment. see last picture at bottom where it say's device this should say Zoom. Look at your settings and make sure they look just like mine, only difference will be instead of Motu your will say Zoom.

Yes there's lots of ways to do that, I'm just assuming the OP has diddly squat for audio gear, but as I said... the latency would make all this pointless. Much better to just share the project like everyone else does. I think it can be done but it would involve very low latency audio devices and A/D - D/A

https://www.voxengo.com/product/span/ Span is free and 1,000's of people use it on the master bus. If you solo a track then you can also use it for troubleshooting or just put it in that tracks effect bin. I also use the Youlean meter https://youlean.co/youlean-loudness-meter/ which is free to try but was well worth the $30 ( on sale) as you can drag and drop any audio file and it will analyze it. A must have tool for working with audio that is going to end up on a delivery system be that a CD or the internet. I can drag and drop finished songs or any tracks even movies.

You would need to use the theme editor which allows custom colours. https://www.cakewalk.com/Documentation?product=Cakewalk&language=3&help=Introduction.27.html I'm not sure where you get it from as it was part of Sonar for me. @scook will know.

Pray tell how are you planning on transmitting his signal to your computers sound cards input jack?

This has nothing to do with Cakewalk. What your asking is more to do with the capture of live streaming audio by your sound card. The problem is this audio is from the output of the system and you need an input. Cakewalk can record inputs but not output unless it is looped back somehow My Motu M4 has loop back so I could do this but it’s rare an on board sound card would have this option The other issue is the latency will be unusable even if you got it to work. Your playback will take about 20 ms to get to your friends house and then their voice will take another 20 ms to return not to mention the real bad fidelity. Best to just use a cloud storage to share the project on line. Cakewalk is free so you can have identical set ups

I see on the support page that it has a driver which no doubt is ASIO make sure you install and choose that in preferences/ audio The zoom should show in all the preferences dialogue boxes Open all the audio settings and check

You had my curiosity. So I tried this and I do not have the resetting issue. The Gain defiantly jumps to 0 after the freeze, but it returns to 4.1 when I un freeze. Set Gain to 4.1 Freeze Synth- note gain is now 0 as you reported Now unfreeze synth and the gain returned to 4.1- it did have to think for about 10 seconds first. Make sure you are using the most updated version of Cakewalk etc. And possibly this only happens with certain VST instruments. So not a Cakewalk issue as far as I can tell. And just so you know, I also froze an Audio track and same results. I also tried 6 different VST instruments including a 32 bit. I even tried adding Automation to a track.

Not free but you might have this one if you own Sonar. It's an odd one because it shows by default on the excluded Plug in list. You have to opene the Plug in Manager and check the Excluded plug in list and enable it. But this is a powerful little processor.