azslow3

"New" features page is not the best to check differences in editions. For Cubase it is https://www.steinberg.net/cubase/compare-editions/ BTW have you managed to scroll down the page I have linked? I mean it gives direct official answer on the question "IS THIS TRUE?" (the subject of this thread) PS. For those who want compare the distribution of features with Ozone 9, f.e.: https://web.archive.org/web/20201112012340/https://www.izotope.com/en/products/ozone/features.html

I am for "backward compatibility" in the number of features for similar named products. In phones, cars, computers and software. But the reality is different. New phones loose connectors, removable batteries and sometimes (in my case) readability of the display (even so it is bigger). Computers no longer have "old" ports and buses. Some audio interfaces are "obsolete" just by one number in one text file of there drivers. Software can loose features. Offline activation can be removed in "minor" version update. I mean we have to accept that, I don't say I find all that good. Note that there are significant changes with iZotope as a company. It is now in Native Instruments "family". Ozone 10 Standard is included into Komplete 14. Some priorities and marketing politic was changes and I guess some "cuts" are the result of these changes. We are on Cakewalk forum, for which significant changes are announced. "Cakewalk Sonar" will have the same name but it will be different product. Some have already noted "CbB is free" and "we have payed for lifetime license". In this thread that is "I could get what I need from Ozone for free (or almost)". Will that change anything? I don't think so. BTW what is included in clearly listed. https://www.izotope.com/en/products/ozone/features.html

Unlike with some other products, it is possible to download and install older versions even after "upgrade". So what's the problem? The company has not packed the features you want into the newer version? Don't buy it or buy different edition. I can complain what my newer mobile does not have in comparison with 6 years old one, from the same company and in the same series. But that is just a bad excuse for not checking what I was going to buy...

I have understood there is a different in "gaming" and "audio" sound cards when by occasion have exchanged SB (ISA) with Gravis Ultrasound in my first PC... Decades later, after connecting M-Audio (Firewire) in addition to SB (PCI), I have realized that Creative gaming cards are still not good for music ?

I don't think without this option REAPER is able significantly beat Cakewalk in number of working plug-ins. As I have mentioned, in my tests some years ago the result was opposite. Since OP has observed the effect (or at least claims he had it...), I assume anticipative processing was on. It can be OP experience the bottleneck from completely different part of the system(s), I don't think he is using Studiocat hardware nor similarly optimized system. But we don't know that. Even so I had no problems with render-ahead (f.e. I don't have UAD), anticipative processing can be switched off per track. Till the effect is using some hardware, inability to work with anticipative processing will probably influence offline rendering as well (from plug-in perspective, except for GUI, that is the same). Also most DAWs support some kind of ahead rendering. I mean such plug-ins will appear "buggy" in many DAWs and particular conditions. I guess in the mean time plug-ins developers are aware.

REAPER has so called "anticipative" processing. During mixing (and on all chains without armed tracks during recording) the processing can be not done in time, it renders ahead 200ms (tunable default). Effectively that means the buffer size is 8820 (at 44.1kHz), independent which buffer setting you have for the audio interface. Cakewalk is traditionally "real-time only". In mixing and recording it does everything with the buffer size of your audio interface. Is the first approach definitively better? It depends. There are consequences. F.e. REAPER quantize PDC (plug-ins own "look-ahead" and so latency, not related to performance, driven by algo the plug-in implements) to the audio interface buffer size (previously for each plug-in, in recent versions per track). Depending from the project, the difference can be significant (usable for recording plug-ins normally have tiny look-ahead, way lower then the buffer size). Also general real-time (recording) can be worse with "mixed" engine (several years ago I have tested the limit with one "heavy" plugin on many tracks, Cakewalk won the test). Also unaware GUIs (almost all...) will "lag" behind the audio for the length of ahead processing, 200ms is noticeable. For 2x2 interfaces, so recording just a tiny part of the project per time, and mixing/mastering, rendering ahead is a good idea. What is the difference when the buffer sizes? Everything still has to be processed "in time", but the computer has more "air". PCs are designed for huge throughput, not for strict real-time. PCs are not DSPs. So without time limit PC can do enormous processing (billions of operations) let say in 1s, but if you want just one arithmetic operation but strictly every 0.1ms, you have to death-optimize the same PC, even so you want "just 10k operations per second". And depending from the hardware and OS it can happened that can't be done at all. How that "in time" is defined? So how much time the DAW+plug-ins really have for the processing? Obviously the buffer size multiplied by sampling rate gives the "time window". But that is not the whole story. The system latency can "eat" a part of that time. In practice the most significant influence has the quality of the audio interface and its drivers. In my tests on absolutely the same system and project, RME driver with 48 samples buffer can happily work while (old) M-Audio with 128 samples cracks and (old) Roland with 128 samples sometimes produced "Stopped audio engine". -------------- How to check the impact of your system latency and audio drivers? Most traditional methods are indirect and they don't give final and easy to interpret answer. Unfortunately Cakewalk also can't report that. In REAPER open Performance Meter and enable all RT settings (not default), including "Hold RT" (but set/unset it to have "current" numbers). Disable anticipative processing. Try playback a single track project, a bit more heavy project, may be even your current Cakewalk project (ReaCWP allows open them in REAPER, not perfect but should be sufficient for performance tests). Is RT longest-block is far from minimum or even over the limit on empty project? There are definitively big problems with system latency, ASIO drivers or both. RT longest-block is ok but not "stable"? There are problems with the system (not optimized). Then check RT longest-block stability in projects with plug-ins. Ignore "initial" jumps, many plug-ins (especially VSTi) lagging at startup. But that can uncover problems with disk related system latency, not visible otherwise. Compare the behavior between different interfaces you have. That can show the "quality" of them (ASIO drivers, not audio...).

I have started "collecting" audio interfaces on e-bay. I mean for someone with Realtek and SB5.1 it is questionable to spend even 100€ for some "device" with unknown outcome (not even 5.1 capable!). It happened M-Audio FireWire Audiophile was around for cheap, and it is still on my desk as "default" interface (after several seconds with Shure headphones, I have recognized that Music Audio Interfaces are not the same as platinum USB cables... each computer upgrade I check if Realtek can do the trick as "default"... but no, at least not yet... lol). Later I took Phonic Firefly. I had to pay more for it, but will way less then for new with 8 channels, able to work standalone and without ground loop issues. There will be someone who wants "professional" FireWire device relatively cheap, as long as: * it works under Window 10/11 (without driver installation tricks) * there are PCIe FireWire cards I mean if FA-101 can be used under Windows 10, it is better mention that when selling on e-bay (or alike platform). And if you have PCIe FireWire which works good (so TI based) and you don't need, it may be worse to include it.

AZController can also send OSC on record/play/mute/etc. toggles. Can be used to auto-change something in OSC capable mixers (Behringer, RME) or do something else wireless way (f.e. NodeMCU+LED matrix). In that direction but without DIY hardware, the approach can be REST commands from custom Control Surface to REST capable smart devices (bulb, power-plug, relay) (f.e. https://www.shelly.cloud/). AZController does not support that at the moment, but if someone really need it, I can add...

Since I work and develop under Linux, I always have X2 in Wine, with several Cakewalk and other plug-ins. The rest in Windows VM. Unfortunately primary music software and hardware is not Linux aware. In reality that could be almost zero effort for plug-ins to support Linux, they all use multi-platform frameworks which are Linux aware. Native Instrument is using Qt (f.e. for Kontakt), which original platform in Linux... But they don't do this. Simply while they can... So for audio recording, Linux is fine. For MIDI it is not worse the trouble. One day that can flip, as I have written most plug-ins can appear under Linux within a day, there are DAWs there, VST3 is also officially supported on Linux. But till that happens, no reason to convince other DAWs. BTW most computers in the word are probably running Linux. Just not end-user PCs, which some people think are "the only computers". But there are servers/farms/grids/clouds, Android devices (phones, TVs, etc.), tiny and embedded devices (f.e. consumer routers), which are technically speaking are computers running Linux. Other UNIX direct successors was no longer popular after related companies disappear (SUN, SGE, DEC), but Apple has decided to go BSD way (probably not liking Linux license). Note that many "own ways" Microsoft was trying with time was also replaced by "normal" UNIX concepts. I mean till now no one has really managed to make something better then this "vintage" concept ?

For me "normal" looks better then "NI"...

Nothing has forced Microsoft to make "MIDI device" to "USB device" mapping tricky. And how they are going to improve that at the end is not really MIDI 1/2/(3...) dependent. Also giving every device an unique Serial is an extra cent (or several) per device. Microsoft and MIDI do not have power to enforce anything. Some companies are ignoring USB specifications even for such devices as USB hubs... who cares...

Some people probably forget (or don't know) that THIS board is visited by many people which was never using (and may be will never use) Cakewalk/Sonar/CbB/Next/Previous/etc. ?

Calling something "Cakewalk Sonar" wakes memories... Especially when it comes to "new"/"innovative"/"multi-platform". 8.5->X1, X3->Platinum, Platinum->Lifetime, "Sonar on Apple", "subscription", DAW-locked plug-ins, etc. With core engine stay the same (obviously it was genius, it has survived two decades...) and "side effects" from mentioned changes. Some previous plans and explanations (partially already mentioned in this thread) concerning CbB already smelled for me. Now there are new "plans", without any real explanations nor details. I guess people who have worked with Cakewalk several decades will continue, no matter what. People for which CbB was just a "free DAW" will switch to another free DAW. And people which have found another "home" will not return back. I mean I don't see a good reason why someone plays "Cakewalk Sonar" card in BandLab game. That term has flavor of "aged, buggy and abandoned" together with added recently "free"...

After Kontakt 7 appears, I have installed Kontakt 7 player. Since then KK loads all instruments in that player instead of Kontakt 6. Now that... Well, I guess I have to upgrade to be less annoyed. Eliminating bad feeling during music creation is worse €100 ?

When the noise is not changing, "by sample" removal works good. Mentioned ReaFIR is probably the best free way for CbB, with real-time spectrum and reduction level adjustment. But find good tutorial how to use it, that is not obvious. For changing noise, payed solutions (f.e. RX) do the job better. In extreme cases, for example when there is background music, the voice should be separated first. F.e. with https://github.com/deezer/spleeter (free). The quality of the result is far from perfect, but better then no processing at all...

For what it is that is not a bad choice. As you can see in referenced thread, msmcleod use and recommend it. He is official control surfaces support for Cakewalk, so you are covered. From functionality point of view it is the same as SL Mixface, just with bigger jogger on the left side and dedicated strip buttons instead of switch mode buttons. Note that with any controller you can use "not official" AZ Controller way. By that you can control whatever is theoretically possible in Cakewalk, for example switching the controller between DAW, plug-in dynamic mapping and MIDI learn in VSTi from the controller itself, assigning arbitrary functions to all controls and combinations, etc. But since I don't have this device there is no ready to use preset.

Original Mackie Control (https://mackie.com/en/products/controllers/mcu-pro-and-xt-pro) has many non-strip buttons (on the right side of strip controls). What these buttons are sending is fixed, but what they supposed to do is particular DAW dependent. There are overlays for each DAW, f.e. https://www.loudtechnologies.eu/shop/en/mackie-mcu-midi-controllers/lexan-overlay/g-10000205 (note there is a separate Cakewalk overlay). Mackie "compatible" devices (NanoKontrol2, Studiologic SL Mixface, etc.) do NOT have all buttons the original device has. They have decided which buttons are most important in particular DAW. "Cubase mode" in Cakewalk surface plug-in tries to use Cubase layout for Mackie Control. So "most important for Cubase" buttons are likely mapped to the corresponding functions in Cakewalk. Original device has motorized faders, encoders and strip displays. Tiny (but in SL Mixface case not really cheap...) "compatible" devices have finite knobs and not motorized short faders. So not only big part of foreseen functionality is not possible at all, what is controllable does not work as foreseen by design. Motorized faders and encoders are always "at correct position", finite controls are not. Depending what you want to do, NanoKontrol or SL Mixface can be reasonable. But be aware what you can control, f.e. compare: https://www.studiologic-music.com/products/mixface/specs/ (note that on the right side of the "DAW" is the WHOLE list what is controllable) with https://www.azslow.com/index.php/topic,604.0.html I mean if you want quickly mix 8 track project by adjusting the volume and pan (and only that), especially if the intention to do this live, SL Mixface will do the trick. If you want control one instrument in a time (MIDI learn inside instrument, rarely switching between instruments), SL Mixface can help as well. For anything else there are better options, even for the same money and size.

Note that all applications and Windows itself should use the same sample rate. Some applications don't show it, but they are obviously also using some value. Interfaces are normally flexible to set any sample rate they support, but once set an application is using the interface it can't be changed. F.e. if you start Firefox and browse YouTube, it opens audio device with some particular sample rate. If you then start any application in any mode with different rate, even so it can "successfully" open the interface with that other rate (interface/drivers dependent), that is not going to work. I have observed many related troubles with that and soft loop-backs. Especially when you have more then one device attached, apps and Windows tend to change there mind about rates, producing all kind of strange results ("child voice" in Zoom, silence, clicks and pops, unusual noise, periodic pauses, etc.). While one interface with own drivers normally "works or doesn't", in the soft loop-back chains "automatic" conversions can produce "partially works" results. Also note that "loopback" feature of most interface is limited. To have the flexibility of soft loop-back inside interface drivers you will need RME...

Start investigation with Windows Resource monitor (resmon). Windows computer starts with several hundreds processes, some of them are "heavy". Many are starting "on there own" later (updates, telemetry, etc.). Don't forget to set "Ultimate" (or at least "High performance") power plan when running DAWs. ASIO buffer affect performance, but in different domain (once you are approaching tiny buffers, the effect is plug-in dependent). Start with 512. Size under 128 may require audio optimization (on system and BIOS level). Sizes under 64 normally require special computer, audio interface and optimization to be useful (for anything except marketing).

"4 months without a hitch" with global echo and build-in MIDI learn (remote control) ??? Man, you are lucky. With that approach I had a problem within the first 4 minutes the first time I have tried to use it, with my first MIDI controller, with one instrument on one track. I have assign the first knob to the track volume. Controlling? Yes. But something is wrong with the sound of the synth... reloaded the project. Everything fine. Controlling the volume - my synth is crazy again. Changed the synth - everything is fine, I can control the volume and that second synth produce expected sound. Problem with the first synth??? Not really... Cakewalk remote control doesn't block assigned MIDI, so it "leaks" to synth

If your keyboard can switch channels easily, you can follow mentioned by John Vere way instead of selecting tracks. That has more chances to work reliable. -------------- At this point I want to mention that following method is NOT for live performance in front of big auditory... At the end I will give an example "why". -------------- I hope you are not using "remote control" way. It is limited... You will have to use Control Surfaces plug-ins approach for anything above mapping particular control in the project to particular hardware element. In "ACT MIDI Controller" that is "Next/Previous Selected Track" assignment to buttons. In "AZ Controller" that is a combination of "Strip Track <Current> -1/+1" and "Function Select strip" Actions. "Strip" Action allows absolute track selection (including by name) and so you can pre-define buttons to select particular tracks instead of moving up and down. "Mackie Control" will not help there with NanoKontrol (real Mackie has "select" buttons) and "Generic surface" doesn't has that function. Some controllers allow computer keyboard shortcuts, so up and down keys. But I don't think Korg has that. -------------- As promised... Put several synthes on several tracks in the project, "Echo" follow focus, so when you select particular track you play one instrument. Now press one note and without releasing it select another track (f.e. by mouse). Press second note... with a bit of luck you will understand what I mean...?

Try https://www.azslow.com/index.php/topic,297.0.html

If you want create a project "on the fly" (so add/replace VST/VSTi or generic way to change the sound), especially without keyboard and mouse, better get host/controller designed for that (f.e. NI Maschine). For just changing the sound on particular track you can use "sub-host" approach (f.e. NI Komplete Kontrol + keyboard). Cakewalk supports MIDI controllers, for direct steering and as a "Control Surface". Using the first option your particular VSTi can be MIDI learned to adjust particular parameters using particular knobs/buttons, some VSTi also allow changing presets this way. As Control Surface (key word for Cakewalk is "ACT"), you can ask buttons/pad on your controller select tracks, control transport, etc. In both cases you have to prepare the project (s) for your live performance, with all assignments already done before.

Note that Sonar (prior Platinum) users can use serial based activation, for corresponding versions of plug-ins (Z3TA+2, DimPro). I don't think there was improvements worse the trouble of "activation". I still remember I put "newer" version of Z3TA+2 on my installation stick, just to find out it requires activation (and as usual in our live, at the moment of installation the Internet was down). I know, it has grace period. But I still prefer to have a "basic" set of music software which I can install without Internet, fortunately Z3TA+2 and DimPro are in that list.

As you could conclude from my previous post, there are such people... At least I (the author of ReaCWP) and the author of AATranslator have tried. Every single tiny parameter and property, including "trivial" (f.e. fader position, timeline position), are not easy to match even with dedicated and analytical effort. In some cases that is at least possible (f.e. fader position), but in most it is not (f.e. an automation of fader, if it has any curves and used interpolations inside DAWs do not match). And so that is the only reasonable way to go ?