John Vere

Where as I agree that having stems is an option it makes for a way more complicated set up. And most people like me who play solo with tracks are also our own song tech. Last thing I need is to be fussing with a mixer. I play the odd show at community events where we get 45 minutes and no sound check. The more complicated your set up the fewer songs you get to play. @TVR PRODUCTIONS What type of music do you play? you sound like all you would really need is drum tracks and possibly some keyboards. So stereo or split stereo might be all you need. My method gives me options of mixing a stereo. Mono or a split stereo Exported wave files. I also play with 1 or 2 more musicians sometimes and we like to downplay the backing tracks. The music is more bluegrass, Folk and Blues oriented. Acoustic guitars, Mandolin and Banjo. So we only really need bass. So I made a split stereo mix with Bass on left and Drums and keyboards in the right. I play Bass on a few songs. Solution. I have a Radial DI box that is made for laptops and iPads. It is a must have and a key feature is it has a Mono toggle for shows with a sound guy. But we mostly do our own sound. We send the Bass to one channel of the mixer and Auxiliary it to a Bass amp. The drums and keyboards go to the PA but we keep that down. It will mostly go to our in ears or floor monitors. Why use a click track?? I’d hate that. The drums are there to keep us in sync but the bass is the driving force and keeps us kicking along. When I play bass I simply mute the Track Bass channel and unmute my bass which is in another channel. My solo act is different depending on event. I do shows with a mix of my originals and covers that fit in. As well as I do parties and dances playing cover tunes. Those I use full stereo mixes and I personally don’t find it that hard to set up that mix in the studio. I’ve never felt the need to mix it live. How many hometown weekend warriors have you heard with a perfect mix? The punters don’t care. They just want to recognize the songs. I always test everything in the studio with the full PA set up exactly like it will be at the gig. I just lower the volume accordingly. And yes, the last thing you want to sound like as a musician is Karaoke. Kill those harmonies. That sounds scary ? The using a multi track recording device is another option and the Zoom L8 is both a mixer and a 8 track recorder. As an experiment I loaded a whole set of stereo tracks to test it out. Compared to using a laptop it’s pretty clunky. I’m used to advancing songs with a USB foot switch. And you have to record everything into it in real time. That’s the deal breaker. I recorded them as I was practicing so time well spent but! And making multi track recordings?!!? But I have them there as emergency back up if laptop died. Possibly there’s multi track recorders that are capable of playing files from a DAW. Otherwise this is totally a step backwards in technology for me.

@User 905133proof that world is flat. As you flew did your drink spill? No because the pilot maintained a level altitude. See flat?

The simplest way to do this is drag and drop your Cakewalk Projects folder to a second drive be it internal or external. This creates a copy that you can now work from. A few other things I would also do. Copy this to a 3 and even a 4th location before you delete the originals on C drive. And open the new copies first to test them as well. I like to have a minimum of 3 copies for back up. Back up storage is dirt cheap. Now in preferences /folder locations change the default project folder location to the new drive so new projects will be stored there. Your start screen will still show the original locations so don’t use that until you have repopulated it with the new location versions. So open the project by either browsing there or by double clicking on them in the new location. If your not short of space on C drive I would leave the original copies there but rename the folder as “Backup CWP Nov2023”.

Be careful with that. It is been said a few times here that too large a buffer size can cause issues in some systems.

My all time favorite is " I've used Sonar/Cakewalk since 1991 and I'm in IT. So I know exactly what I'm doing here" Seem in my experience being in IT doesn't help much when you use a DAW.

How 'd you guess! It's all about comprehension and learning disabilities because a lot of us creative people are ADHD myself included. I found out that many people only watch tutorials in fast motion. This one lady wrote me claiming that , gee, I had to actually watch your tutorial in real time because you don't bullst around like the others so I kept missing stuff. I'm also guilty of clicking to the 3 minute mark on some videos just to get to the lesson.

Good find on that link, And that's first thing I would have done was google the question. Focusrite has excellent support and documentation. It might be if phantom power is on and needed for the mike. A powered hub might be needed. Laptops USB systems are not always very powerful.

Yep. Sure fire way to make it happen. During playback your CPU is distracted and busy. Pretty easy for it to miss a 1,000 bits of info that happens when you drag midi blobs around.

Flat earth is real!

Forget it Not my problem Eh!

Funny as we just closed off another 39 reply 2 page thread where the answer to the OP’s issue was in the very first reply.

You mean buffer settings. Lower buffer settings reduce Round trip latency. But that’s nothing to do with also increasing them to stop drop outs and static. Latency is only noticeable when you turn on input echo when recording audio or midi. Not during playback. This why interfaces have direct monitoring so latency is not a factor in tracking audio. Round trip latency is more or less invisible to anyone who records, edits and mixes with a DAW. But buffer setting are important and performance is based on the quality of the system. A powerful computer with lots of cores and RAM will run at a lower buffer than a not so powerful one. Then some ASIO drivers are better than others like RME, and that also helps prevent dropping out at lower buffer settings. But as projects grow and certain plug ins are added that starts to stress the system and that might require increasing the buffers during mixing to stop glitching. But increasing buffer settings and then trying to record midi you might hear the latency so we then just globally bypass the effects and 99% of the time this fixes it. Then exporting a mix is all in the box and has nothing to do with buffer setting or your interfaces drivers. It just goes faster on a more powerful machine. I always set mine at 256 and forget about them. But that said my now retired Motu M4 was on the edge at 256 so I had to bump it up. I now have a Zoom L8 and it is on the edge at 128 so I’m back to smooth sailing at 256 again .

I’ve been on a quest for a ride cymbal that actually sounds like a ride cymbal and not a crash. I think I’ll just record my own someday. I did this about 15 years ago but I lost the folder in a hard drive failing. I recorded my whole kit and it was cool to play it using Session Drummer.

Ya I mention it when I get a chance because it’s free so no harm in trying. I had to watch a video to get it working but have no clue which one that was, but yes it’s a strange duck. Good news is once you get it working it’s very easy to use. Try different streaming settings if your getting glitches.

Don’t do anything in Cakewalk. Just choose Mono when you export. Question? What are you using for playback at the gig that requires a mono MP3 file? Just wondering at that choice?

Sure looks good to me. That’s first class support I’d say. No harm in re installing it will either work or tell you it’s already installed. People are always uninstalling and I don’t think it’s ever been necessary.

I have in order of purchase M Audio Fast Track Pro, Tascam us1641 , Focusrite Scarlett 6i6, Motu M4 and the latest a Zoom L8. These are all what we would call entry level consumer gear. Anyhow shopping for an interface based on thinking there will be a big difference in sound for $200 is a very questionable exercise. They are all about the same. And ask 100 people you will get a 100 answers as it’s subjective. You are better off making a list of features you want and compare that. I think you’ll be disappointed in all offerings that are $200 range and the Behringer might be better than some of the others. That’s one that has real ASIO drivers right? Out of all my interfaces the sound quality has always been indistinguishable but the features make a huge difference to my work flow. I Scraped the Tascam because back then the driver sucked. They fixed it after I bought the Scarlett. I scraped the Focusrite because I hate the monitoring set up. I then scraped the Motu because it craps out randomly( Due to bus power?) So far the Zoom L8 outshines them all because it has 10x the features and everything on my list is there in a neat small package. Zoom was not ever on my radar until I started shopping this fall. I almost skipped it over because of its past reputation for bad drivers. Oh and it has a HiZ input that is working really well with my guitars and bass. It’s double the price of your budget but that’s my point I’m making. Anyhow make that list first. How many inputs do you need and are they multi purpose? How many outs How many separate headphone outs. Can you adjust the mix of the direct sound with playback using the front panel? Is it bus powered and are you aware of the issue that can cause ? Does the company update its drivers? Do they have good service? Do you need Loopback?

@Craig Anderton Yes it is a big concern and I think one of the reasons I chose-1.0 db was years ago there were many debates on this topic. It was one of those everybody had their opinions endless tail chase debates kinda like sample rate or analog vs digital debates. So I took the middle ground. Truth is I had been normalizing all my wave files to -0.2 db using Wave Lab for a long time. This was in the CD and DAT days. After I learned how “wrong” this was I took a close listen to those files and never really heard anything wrong? Much later on I dropped them in You Lean and most were dead on 0 db. But nothing was a +db. And then began my using the BT brickwall set at -1.0 db. Later using You Lean I learned that most of those masters were very close to -0.1 db? I then tested a dozen or more free Limiters and the Loudmax and Boost11 won the shoot out. Set at -1.0 Loud max at worst is -0.8. I apologize to Richard for our geeky side track but this is what we do. For your backing tracks the peak limit is not as mission critical but the LUFS definitely are . This latest batch sound way better than my last go round pre Covid. And people noticed. I’m still performing at 70 and I need all the help I can get so perfect backing tracks are worth the extra time and effort. Nothing I can do about arthritis and my guitar playing but the rest of the band is in top form now. And they don’t miss practice or get drunk at gigs. And oddly enough I get paid the same as a 3 piece band. But I do share this with the band and I’m going to treat them to a new computer.

I did already.

We are not allowed to discuss this topic. There are at least 100 pages already here you can read at your leisure.

Cakewalk is real good at taking anything you throw at it. If a project is 44.1. And you drag and drop a 48 or even a MP3 audio file it automatically converts the audio to 44.1.. Of note is the bit depth stays the same as original as Cakewalk can have audio files at any bit depth, but sample rate has to be the same.

The fact that WASAPI Shard mode doesn't offer buffer adjustments is 100% on Window and your Audio systems end. It has nothing to do with Cakewalk. This is why if you are using on board audio and want ASIO type features and performance you use WASAPI Exclusive mode. That is as close as it will get. ASIO4all is really only WDM mode in a 3rd party wrapper. This is why Cakewalk staff do not recommend it. Any Vendor that does is really out of touch with Windows 10 and 11 audio realities and I would personally avoid their products. Asio4all= WDM mode is the old Windows code and worse performance than WASAPI shared. Sure it might work but you'll just keep digging a deeper hole. Reaper uses a completely different audio engine than many DAWs do.

Another reason for not using MP3 files unless needed for the internet. But that said I do bulk convert in Gold Wave to MP3 that are 48 / 320 kbps. I upload these to Sound Click so I can listen on my Cell phone via Headphones. I can't hear any distortion so possibly it depends on the convertor and the sample /bit rates. Example I never use Cakewalks MP3 export. @RICHARD HUTCHINS I couldn't find the exact video I think I removed it last year, but this video has a lot of the same concepts demoed.

I’m your man! I just redid over 200 tracks last winter. The ultimate secret weapon is the You Lean Loudness meter. To save a lot of time get the paid version. This allows you to get a reading in seconds. The free version requires playing the file in real time. The reason i say this is your most important statistic is what the LUFS are for the song. This is the all important perceived volume. Exactly what you are trying to do. So I choose-14 LUFS at -1.0 db peak level as my target. This alone will bring your songs into the ballpark. But there’s a little more to it as the songs need to be mixed with things like drums and bass always being proportional. My solution to this is I always use the same exact instruments and pre sets. Ample P Bass and Addictive Drums. I also always use the exact same velocities for these. Next I use 3 or 4 Sub buses. Bass, Drums and Keyboards. I rarely use VST guitar or Horns but if so it gets a bus. Next I have the Loudmax brick wall limiter on the master and the You Lean meter underneath. I set the Loud Max at -1.0 db. Because the bass is the same VST set with same volume and velocity it is always seems to be at about-4 db. Perfect. The drums are about the same. Now keyboard stuff is always different so I mix it until it is where I want it. Now check the You Lean meter and it’s usually showing I’m good. If it’s too hot and the loud max is showing gain reduction I back off on the keyboard first. I export these at 48/16 wave and drag them into YouLean meter to test. The end result is I never need to adjust volume at gigs . It was time well spent. I made a video once I’ll see if I can find it.

Just a question but why do you need to use 44.1? Is that the delivery format? I only ask because if you’re doing it to decrease the file size it’s not a heck of a lot. Decreasing the bit depth will make more of a difference. You should be able to open either format but as David said some interfaces don’t do this very nicely. My Motu is one of them. My other interfaces no problem. Always check in the transport module to make sure the Sample and Bit rates are what you wanted before you record.