msmcleod

Try disabling the "HD Audio Driver for Display Audio" driver within Windows' device manager. This is the audio device that is used when connecting to an HDMI TV - on some machines it's been known to cause issues with both stability and system performance.

Exporting each project as a stereo file and then creating a master project from them would be my preferred method - that's assuming that you've finished editing the project. The alternative is to combine all of the projects into one, and although you can use track templates for this, this is only really to get around issues with inter-project copying/pasting. The basic premise is: For each project, select all then save as a track template - in other words, each track template will contain ALL of the tracks of each project. So you end up a single track template for each project. Ensure "Allow Only One Open Project at a Time" is unchecked within Preferences->File->Advanced. Create your new project Set the now time to 1:01:000 Insert track template 1 at the bottom your new project Open project 1, then use Window->Tile in Columns to put your new project and project 1 side by side. Select all in Project 1, then Copy Special, making sure everything is checked In your new project, ensure the top track of the template is the active track Paste Special Close Project 1, and delete any tracks you don't need Within your new project, set the now time to the end of the project (where the next part will be pasted) Go back to step 5 and repeat for each project The downside of doing this is that with all of the extra tracks, you're potentially putting a huge extra load on your CPU. You can however get the best of both worlds by bouncing each part to an audio track, then archiving the original tracks - that way all you're doing is playing a single audio track for each part. If you need to edit, delete the bounced audio clip, unarchive, do your edits, then re-bounce.

If you're using WASAPI Exclusive it'll take exclusive control of the audio driver. Setting it to WASAPI Shared will share it with other apps. Alternatively, check "Suspend Audio Engine When Sonar Is Not in Focus" within Preferences->Audio->Playback and Recording.

You shouldn't lose any settings when installing an earlier product, as all of the config files & registry settings go into their own directory. Even uninstalling a product leaves your config files & registry settings intact - this is deliberate, so you can uninstall / re-install without losing anything. There are however some shared components that all products will share - examples include Pro Channel modules, internal synths (e.g. TTS-1), and utility apps such as the plugin scanner. Normally the installers won't overwrite a newer version of a common component, but there is always a danger that it might. This is why you should re-install the latest product after installing something like SONAR Platinum... definitely re-install CbB or Sonar after installing something as old as SONAR 3! As far as backing up your settings, in the Sonar Preferences there is a "Backup / Restore Settings" section under Customization. There you can save your complete configuration as a single file, and restore it as necessary. You can even have different configurations and use desktop shortcuts to launch Sonar with the configuration you require. Instructions on doing this is included in the latest help documentation.

Streamworks Audio's SWA Complete SONAR X2 is old, but covers 90%+ of what Cakewalk does: @abacab has kindly listed a description and links to each chapter in this post:

All you really need is a decent (i.e. fast) PC / laptop, an audio interface and optionally a mic if you're recording acoustic instruments or vocals. Personally, I'd avoid laptops as the more powerful ones sound like jet engines when the fan kicks in, but this may not be an issue depending on what/how you're recording. If you don't use many plugins, you can get away with a far more modest PC setup. If you intend to track using effects though, you'll want a fast PC and an interface with a low round-trip latency. Latency can be reduced by lowering the ASIO buffer size, but you need a fast enough PC to cope with this. If you're using software synths, you definitely want a fast PC. Most audio interfaces have very good pre-amps nowadays. The Focusrite Scarlett range (e.g. Solo / 2i2) are excellent. I find the 1st gen Scarletts more than good enough for my needs, but the later ones have improved preamps. I've heard you should avoid the 1st gen interfaces if you're using the HiZ input for DI'ing guitars (which I never use). You can pick up 2nd/3rd gen interfaces at very reasonable prices second hand. The PreSonus Audiobox 96 is another good contender - fantastic price, great quality, includes a MIDI interface too - but it does have higher round-trip latency than similar interfaces from other vendors. Bear this in mind if you intend to track guitar using VST effects, but it may not be an issue at lower buffer sizes. As far as mics are concerned, a Shure SM57 will cover most bases (or a Shure SM58 for vocals). The SM57 is certainly not the best, but it's a good quality mic that does the job, and you can do a lot with EQ to get the sound you want. If you're recording acoustic guitar, a condenser mic might be more suitable (e.g. AT2020). The Samson C01 is a good all rounder budget condenser - it's not harsh in the top end like a lot of budget condensers, but it does suffer from a bit of hiss.

Sounds like the 2nd graphics card is using the same IRQ as either your audio device or your disk controller. In the old days, you'd just move it to another PCI slot... but nowadays slots are limited. Check your motherboard documentation and find out which slots / ports share IRQ lanes, and (if you can) pick a combination that will mean that two high traffic devices aren't sharing the same IRQ. FWIW - this is why for some users, disabling the network adapter can improve audio performance. Don't just disable stuff though - check the motherboard docs so you can be deterministic about it.

When you're recording automation, you're shown a preview of the automation - more to give you some confidence that the automation is being recorded than anything else. The automation points are then "smoothed" after recording, picking curves that best fit the recorded points. In the new Sonar you have more control over this process: in the "Automation Smoothing" menu under Track View Options, you can choose: Smoothing: same behaviour as CbB / SONAR Platinum - This reduces the number of recorded automation points by picking curves that best fit the recorded points. Useful if you expect to manually edit the automation afterwards, as you end up with a smoother automation lane than hundreds of automation nodes. As Recorded (Linear): preserves all the recorded points received from the controller as individual nodes, and joins them with linear lines. Use this when you need a smooth transition between points, and accuracy is more important than editing afterwards (there will be far more individual nodes than the smoothing behaviour). As Recorded (Jumps): similar to above.. it preserves all the recorded points received from the controller as individual nodes, but doesn't join them. Use this when you don't want a transition between the points - i.e. when recording "mode" type parameters such as synth oscillator waveform, EQ modes or on/off parameters.

The reason RealVNC costs is due to the encryption it offers - so unless you intend to connect remotely over the internet (as opposed to over a local network), you can use any number of free VNC servers/clients (e.g. TightVNC). Even Microsoft's Remote Desktop (which is built into Windows), will do the same job. If you're only ever connecting to devices across your own home network, save yourself some money and use the free tools. FWIW - I've got RealVNC set up to listen on two ports: one has encryption for remote access, the other has no encryption but is firewalled to the outside world at my router. Because the VNC protocol is universal, I can use any VNC client to access my PC on the unencypted port, including those on Mac, iPhone and Linux.

This is a browser thing - click on the "Insert Link" button - you can paste the URL and the text that is shown for the link:

As a starting point, I'd recommend these books: https://www.amazon.co.uk/Designing-Audio-Effect-Plugins-Theory/dp/1138591939 https://www.amazon.co.uk/Designing-Software-Synthesizer-Plugins-Audio/dp/0367510464 The software that comes with the book creates a lot of the boiler-plate code needed for creating a VST3 plugin - all you need to do is fill in the blanks. Even if the UI elements it aren't quite what you want, it'll at get you to the stage of being able to test your DSP code very quickly. You could then use something like JUCE to create a more elegant UI and cut/paste the DSP code into your JUCE project. Of course, these books only cover VST / VSTi plugins. If you need to control the workflow within Cakewalk/Sonar itself, the only real way to do this is via the Cakewalk Control Surface SDK. This will give you programmatic access to running Cakewalk commands: https://github.com/Cakewalk/Cakewalk-Control-Surface-SDK

This is what I use for SF 12:

In the new Sonar, you can set the Start Marker to where you want the music to start. It doesn't change the bar numbers, but it'll save you having to fast forward. In Cakewalk by BandLab, one workaround is to enable looping, setting the loop start to where you want the music to start, and the loop end way past the end of the project. Pressing W will toggle between 1:01:000 and the loop start, so it's easy to back to where you want , and there's little danger of it looping back if it's far enough beyond the project end. Another method is to use markers or arranger sections, and then use the marker view or arranger inspector to jump to the marker/arranger section that matches the point in the project you want to jump to.

This isn't true. As long as you've got the latest security updates to Windows 7 (with TLS1.2 enabled) and the latest CbB, it should activate fine. You'll get a warning on installation, but that's all it is - a warning. I'm running the latest CbB on my Windows 7 boot with no issues. FWIW - I've tried the new Sonar on Windows 7, and it doesn't work properly. Menus are broken, drag/drop from the browser is broken etc.. It's not surprising - Sonar is using Windows API's that are only available in Windows 10 and above.

Network transfer is way too slow for me on the Pi - I'm lucky to get 11Mb/s on the PI vs 16GB/s with a portable drive.

That used to be the case, but the difference in energy costs was surprising. Turning everything off has reduced my fuel bills to around 1/3rd of what they once were. The only devices I leave on now are a couple of Raspberry Pi's and a MeLe silent PC, both of which are very light on power.

True... but can be a PITA when it comes to backing up! I used to be able to backup my whole system in an hour... it now takes over 6 hours, and I need a monster of a backup drive for the drive images.

That's strange... I don't recall any sample rate restrictions for Sonarworks. I've got the headphone release only, so there was no measurements involved. I've got ARC 2, though which has a 48Khz restriction for measurement setup only, but after the measurement was taken you could use any sample rate you wanted.

IIRC these are the project preview files (used in the start screen). If "Include Project Preview" is unchecked in the export dialog, they won't be generated.

As @User 905133 says, it's almost certainly from old player-pianos: When I used Music-X on the Amiga back in the late 80's, it had a piano roll editor:

I've had this before - quite recently actually. I fixed it by opening Native Access, then the Kontakt 7 standalone - everything was fine after that. FWIW - I always find it's good practice to open the standalone version of any app that comes in both standalone & VST format, first. Some take a while to update their internal databases/phone home, and this can confuse the VST scanners of DAW's.

In Sonar, not one click but two:

It's worth taking a look at this article explaining how the P/E core scheduling works on Windows 10 & Windows 11: https://aloiskraus.wordpress.com/2024/02/08/hybrid-cpu-performance-on-windows-10-and-11/ TLDR: If the process is running at a priority of normal or above, then Windows will choose the performance cores first, then fall back to the efficiency cores. If the process is running below normal, it'll pick the efficiency cores first. The Audio engine in Sonar/CbB runs at a fairly high priority (higher if MMCSS is checked in preferences, which it is by default), so Windows should be giving priority to performance cores. If e-cores do end up used for audio processing you could experience drop-outs if the e-cores fail to process audio in time. In essence they become the weakest link in the chain. In saying that, the e-core max turbo frequency is 4.30 GHz on a 13000K, and 4.5GHz on the i9-14900KS which should be ample for even the most demanding project. Of course, the fix is just to up your ASIO buffer size if/when this happens. The only time having a higher ASIO buffer could cause issues, is when you need to re-record or overdub a track later on in the mixing process. There are various workarounds for this including freezing tracks, turning off effects, or bouncing all the other tracks to a single audio track and temporarily archiving those other tracks while you track.

You don't have to use the headphone correction - you can enable the room simulation on its own. The adjusted frequency response won't be 100%, but it will give you the room imaging/cross talk etc. Alternatively you could download (or measure) the frequency response of your current headphones and chose a headphone model with the most similar frequency response. Lastly, you could try contacting HorNET so see if they'd include your headphones in their model list.

I've got it, but haven't really used it much. Seems to do what it says on the tin.