John Vere

All good here now, Thanks for the very, very quick fix. I'll watch for the official release for my main DAW now.

Not sure what your getting at because you don’t mention which playback format or device you are targeting. You can make a CD and that’s an album. For mp3 files I often put the songs track number first in its title 1-first song 2- second song Then most MP3 players will play in order.

Life is better when you use a proper audio interface. That is the only way you'll get the true performance of ASIO. On Board audio is OK for editing and playing audio. If your getting dropouts possibly your system was not optimized for audio. Run this utility to test- https://www.resplendence.com/latencymon But still,, even if you do that, you should purchase an audio interface. This video explains why- https://youtu.be/Avtw7dOb0fM Here is a blurb I made about buying an audio interface: 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, 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 all ¼” jacks Balanced? Some have RCA jacks in Parallel with main outputs. Is there a true stereo input pair? Some don’t have a matched set of inputs which sucks for recording stereo devices like outboard synths and drum machines. Are there peak level meters or just a little LED for each input? Do all inputs have a peak indicator? Are there channel Insert jacks? Are there separate controls for 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. How many Headphone jacks? A level for each? This is also important if you work with other musicians. 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? 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 live streaming, screen captures, stand alone VST recording or wish to record any playback from outside your DAW. What are the Round Trip Latency (RTL) specs? Do you need low RTL for real time processing? Low RTL is going to be at a higher price point. Most interfaces under $600 will have hidden buffers etc. Zero Latency monitoring is not the same as RTL. All interfaces have some latency. Zero latency is just marketing hype for monitoring directly from the interface. And most important of all, Does it have top notch ASIO drivers for your OS. Don’t buy any interface that say’s “class compliant driver” that will work for a Mac, but not very well with a PC. 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. See if the drivers and manuals are easy to get at and 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. Check for pricing on sites like Sweetwater or Musicians Friend.

I can just see them scratching there heads now! But this all just goes to show you how important the ASIO timing master and calculations are. I'm glad I didn't update my main DAW as I'm right in the middle of a song and it will need punches like always when I get going. This would drive you bonkers if it was how a DAW normally worked. This is all normal behavior if your not using an ASIO driver. I remember my Sound Blaster days and this was about it.

If Zoom is like OSB studio what I had to do was set Cakewalk to WASAPI shared mode to get it to work. BUT- and I see you are lucky because your Focusrite seems to have a loopback channel. Once I figured that out I now have both my live mike and Cakewalks playback into OSB and I can use ASIO. Not sure why this works but otherwise like you my mike worked but no Cakewalk playback. I had to choose the Loopback channel as one of the input sources in OSB. By the way, your buffer setting seems a little high. I use 256. Also make sure all your settings are using the same sample rate. either 44.1 or 48.. even check in Windows sound settings because that is what Zoom probably uses. Oh and get rid of asio4all. You really don't need it because we now have WASAPI as a better alternative for on board audio. Always use ASIO with the Focusrite and use WASAPI shared with your on board if you turn off the Focusrite. asio4all can sometime interfere with stuff like this.

I can see that what is happening and I have also complained about this. They have made the basic templates with most of the features missing. They say it's too confusing for new users,, I say too bad, they can suck it up. So what you have to do every time is change the Track Control from either workspace or Custom to ALL. My workaround is I opened the Track control and changed all the options to always use "select all controls"

I just noticed there's another thread regarding this. Anyhow yes- the punch in is visibly early on the timeline. Even without the audio the punch in point is early as you can see here. l

LoudMax- I just love this simple little guy. It seems to use low resources too. So you can add it to any track that you feel needs some processing. Mostly I like that it is a simple one control brick wall limiter. I even use it on the audio output of my movie editing software. It seems to be a colorless compressor and doesn't sound compressed at all when you push it. And so simple to set up. Just set the OUT to say -8.0db. Then the threshold you push until you hear the tracks volume increasing. You'll find the compressor kicks in at whatever the tracks peak level was. So say the track was originally peaking at -10.0db and you want 2 db more, just set the threshold at -12.0 db The compressor will even the track out and the limiter will keep it under -8.0 db I've yet to see that it fail in this important task. If I set the "out" at my desired limit it shows this on the You Lean Loudness meter. I was mostly using the BT Brick wall but I was seeing different results often. I'd set the BT at -1.0 db on a master bus and both Wave Lab and You Lean were reporting -0.7db. https://loudmax.blogspot.com/2021/02/loudmax-plugin-v138-released.html The You Lean Loudness Meter- If you buy the Pro Version of this for $ 47.50 Can. you get that drag and drop feature which is a huge time saver. In seconds you know the total true peak level and most important the LUFS of any audio track. Best money I ever spent on a plug in. https://youlean.co/youlean-loudness-meter/

Here's a screen shot of a typical finished song. I choose one that I have absolutely no sub busses on for clarity. You can see that most faders are at unity with the exception of the guitars which have automation and are turned down. Things like the bass and drums I have used the gain control to lower them so peaks are around -10.0 and you see the vocals at around -8.0 . Not shown is the Pro Channel where there will be some EQ and compression.

If this is the vocal track your refering as to being too quiet then there's a bit of a mystery because -2.9 is dangerously close to clipping. It would be nice to see a little more detail in the screen shot like all the track settings, faders and outputs etc. Drag out the track inspector part so it shows all the details and select ALL I think your on Custom which hides important stuff. . And at this point there's not a lot of other tracks playing. A shot at the busiest part of the song would be nice. But right off, the one track with piano is showing -5.0 peak and the drums are also -2.0 which is also loud. When you said you were recording at -18.0 I was thinking you were a bit to low as I aim for around -10.0 for instruments and -8.0 for vocals. The video I made is about turning up levels but I'm thinking you need to defiantly turn the instruments down if they are all peaking between -5 and -2 db. In my video I demonstrate using the Gain to turn things up,,, just do the opposite and turn down your instruments so they are closer to -8.0 or even -10.0

Looking at the screen shot the blue track is showing 2.4 peak level. Is that the vocal?

Watch the video before you delete anything. I give you 4 ways to fix this. Good luck

No because I’m testing this on my office computer as is my usual habit. I don’t commit anything to my main daw unless I’m dead sure of it. So I don’t have any recording stuff there other than my interface. I was also thinking it’s hard to have an accurate reading because now a human is involved ? But I’ll grab a guitar and plug it in later once I get my 4 th coffee I gather you are referring to an auto punch And I’ll add I’m pretty impressed with how quickly staff responded to this. Try that with any other daw

Mine is showing good now on the loopback. Thanks.

Thinking about this and the fact that there isn’t any GM ready VST’s out there. Well I do have Steinbergs Halcyon which has real good quality sounds but that’s difficult to use in Cakewalk But seems TTS-1 is an important part of Cakewalk. How hard would it be to update it? Is it impossible? What would be cool would be if it became a VST and used basic samples. They could integrate SI instruments. Just think how great it would be to NOT have to swap out most of the sounds.

I didn't read all of this but I made 2 videos that show you how to use 2 instances to get 8 audio tracks. in my example I output the first 4 as instruments and the second instance I use as a 4 track drum module. I was having crashes back a few years ago but it seems Cakewalk updates have eliminated that for me. I don't really use it on any finished songs other than I might steal the brushes or percussion. I'm not a fan of it's lo fi sounds. But it's a super important tool for pre viewing midi files and getting a start. Oh and way back someone mentioned Xpand 2 as a replacement. Ya, Xpand 2 is not GM so will not play midi files out of the box. But I will say it's what I do use to replace TTS-1 for horns, woodwinds and percussion.

I think the OP is being confused because Google search often brings up old web pages and they really need to delete those as they are misleading. Here's a example. I simply typed in Sonar Home Studio to Google and this is what you get... It say's Buy Now-

SI instruments are a separate download than the main Program. It is found in the install ad ons tab

And believe it or not there is a lot of music and projects recorded in Cakewalk that would not use drums. I often use Cakewalk to record dialog for tutorials. I also record Bluegrass. Sorry, Cakewalk is not a Cell phone App. It's a fully featured DAW that can do just about everything Audio and Midi.

But I think it might be inconsistent. I'm sure they'll fix it. There's seems to be a few threads reporting issues already. Lets draw out some staff and see what they say. @Jesse Jost @Noel Borthwick

This was interesting as well. I changed buffer size which in past experiences has never had any bearing on loopback results at all. But as you know it does change the reported RTL in SYnc and Cashing settings. I got different results and different settings. At 64 ms buffer it was late only 500 samples now. But at 560 ms buffer it was late by over 1,000 samples. So it would seem the calculation is totally broken. top track is frozen drums, / 2nd track build 98/ 3rd track 256 buffer / 4th track 64 buffer/ 5th track 560 buffer

@DeeringAmps and @Base 57 So I ran the loopback test using my Scarlett 6i6 ASIO mode with version 2021.01 build 98 and as expected it was right on as seen in track 3. Up updated to 2021. 04 build 144 and now it is late by what appears to be 800 samples. see track 4 So defiantly don't update to the new release folks until they fix this. Its a big one. How the heck did they break that?? And I was going to update soon. Thanks for pointing this out most of the people using this software would not even notice.

Glad I didn't update yet. This is a serious issue. I'll update one of my computers and test later tonight.

Hasn’t there alway been a reason we are told never put data at ground zero? I think when you hit play there is always a delay before stuff happens so we are advised to start at at least measure 1 and many use 2. A long standing feature request is to have a negative measure at the start because of this. It would be used for count in and pick up notes And if you need a good tool to perform the loop back try latency monitor https://www.resplendence.com/latencymon