msmcleod

As @Starship Krupa said, we really need to know more about your workflow and what you need out of the interface. If you're only ever going to record one or two audio tracks at a time, then something like a Scarlett 2i2 will do. If you need MIDI or an extra couple of audio inputs / outputs then maybe a Scarlett 6i6 will do (tip - if you ever think you'll need to use a re-amp box or external effects, then you'll need spare inputs & outputs). If you think you'll ever record live drums, then you'll need something that will take at least 8 inputs - maybe more, depending on how you want to mic it. The Scarlett 18i8 or 18i20 might be good choices in that case. Both have 8 analog inputs (plus a stereo SPDIF) and an ADAT in for a further 8 inputs. The 18i20 also has an ADAT out for 8 additional outputs. Those are just some of the options from Focusrite - most other manufacturers have similar options (e.g. Tascam, Audient, Presonus etc), and some come with extra features at the lower end of the cost scale - e.g. the Presonus Audiobox USB96 is one of the cheapest interfaces available, with 2 inputs/output (like the Scarlett 2i2), but it also has MIDI in & out. If you're used to using a mixer, then the Tascam Model 12 is a great choice (there's also the Model 16 / Model 24, but they are VERY different beasts and in some ways far more limiting with regards to routing - e.g. you can't apply the mixer's EQ / compression / fader levels to what you record on the 16 or 24, but you can on the 12). Zoom do some good interfaces too, such as the LiveTrak L8 / L20, although I would say that although they are great interfaces, their mixing duties are primarily focused on live use - i.e. you use the mixer for your live mix, and what is recorded isn't affected the mixer's EQ/faders etc. This is also true of the Tascam Model 16 & 24, and the Soundcraft MTK series. If you do go for a combined mixer/audio interface, please ensure the following: It exposes individual track outputs to the audio interface, not just a stereo output... otherwise you're back to the same issues as you've got with the Alesis. It has a decent ASIO driver, and doesn't rely on something like ASIO4ALL. You're happy with where the signal to the audio interface is picked up from - i.e. is it only just after the preamps, or can you apply the EQ/Compressor/Fader from the mixer too? If you're intending to use condenser microphones, does the mixer have 48V phantom power?

There's a difference between track focus and selection. The answer is simple: If you don't want the clips selected, then don't click the clips. Clicking anywhere in the track header (apart from the track number) will focus it without changing selection. I usually click the track icon, track name or the blank space underneath the MSR buttons. Clicking on the track number however, will select the track and all clips within it. This is actually my preferred way of selecting all the clips in a track - especially when used in conjunction with the PRV, as it guarantees that they're all selected - clicking on a single clip may not select others if they're hidden away in a take lane and further down in the project (unless it's the comp clip you're clicking - that's ok). If you've been used to always clicking clips, I appreciate this may take some undoing of muscle memory!

Add the task to your project template then. If the task is created as follows: - Nothing selected ( so CTRL + SHIFT + A before you open the export audio dialog) - Source Category = Entire Mix - Range = Entire Project ...then the task will work regardless of changing tracks/buses. Double clicking the task will recall it allowing you to edit it / create new tasks as required.

I'd add SynthMaster 2.9 alongside Omnisphere - it's a killer synth, with a huge library but covers a different set of sounds. If I was to choose only one purchase though, I'm pretty sure I could survive with only the Korg Collection 4 - it pretty much covers all the bases.

Correct.

You've got the clock source the wrong way around. If the Evo SP8 is the ADAT OUT going into the Focusrite ADAT IN, then the Focusrite has to have its clock source set to ADAT.

Check your Clock settings within Preferences->Project->Clock. Your source should be set to "Audio" in order for it to use your Focusrite for timing:

It's almost certainly down to one of your synths - probably a sample-based synth with a huge sample library. Most "library" based synths have the ability within the plugin itself to limit the amount of memory used and/or load samples on demand, so samples are loaded on when they are needed rather than loading everything into RAM. Loading samples on demand will not only save RAM when using the synths, but will speed up project loading time significantly. The downside is that there may be a delay in loading the samples on first playback - i.e. you may need to play your project once all of the way through to ensure every sample used is loaded into RAM. Kontakt for example can not only be configured to limit memory/load on demand, you can also "purge" the samples, effectively emptying the memory of any unused samples. You'll need to consult the documentation for the synths in question to see where these options are.

You could always use BandLab itself? Cakewalk can upload your ideas as a new BandLab project, and you can invite others to add to it if you want. BandLab will run on most browsers, and also iPhone, iPad & Android devices. Best of all, it's free.

Just noticed your project name... just a hunch, but maybe the spaces are confusing things (reverse is a very old function) - try saving the project using underlines instead.

Looks like the FX bin is bypassed - click the "power" button in the top left of the FX bin.

Yes, the list is fixed. However, included in the list are virtual key presses that can be sent back to Cakewalk/Sonar. If these are mapped to something else within keyboard shortcuts, then those bindings will be executed. Using this method you can get the function keys to run CAL scripts or open Studioware panels.

Ok, look at the MIDI messages that are being sent by the Icon, then look at the corresponding message in the document I posted a link to. I suspect the messages being sent are those of the function buttons. They can be remapped as required using the Mackie Control Surface Dialog:

The MCU protocol uses MIDI note on/off messages for button presses. There are standard mappings for these buttons for the various DAW's, and are usually hard-coded in firmware for non-Mackie control surfaces. Cakewalk/Sonar supports the standard Cakewalk/SONAR mapping, Cubase and the "Universal" mapping. Cakewalk's Mackie Control Surface dialog allows you to map the 10 function buttons to various Cakewalk commands/functions, but beyond that there's not much more configuration. If you need more extensive flexibility, you could look at @azslow3's AZController. The level of customization is extremely flexible, but can be complex to set up - especially if you're unaware of the exact messages being sent by your control surface. Anyhow, in answer to your question, Cakewalk's mapping of messages are listed here: https://msmcleod.co.uk/cakewalk/Cakewalk_MCU.pdf

sounds like a plan

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: