LarsF

I used chord track in Cubase and StudioOne. I never use stretching and that kind of stuff, to switch key on full project or anything. Daw is just a holder or recorded clips, some plugins, and automation - for me. Chords are just for visual presentation for guitar or keyboard as I am recording. Projects evolve over many years for me - so kind of forget harmonies when coming back to a project and arrange a bit further. And moving projects from StudioOne last I missed chords I had access to a lot. So I used conversion of audio clips to chords as well as midi clips of course. Sometimes do free playing and record and would be smooth to get chords up there. I can also from this see good use for staff view - to select guitar chord diagrams for sheet music. Almost automatic, or get a suggestion and you can browse and select another or get another inversion etc.

Many thanks. I found that punch-in recording always play back previous loop as you record next - not usable. Tested both to check Mutes takes and unchecked too. Doing normal loop only, at least I can use since not hearing previous loop while recording. But original issue remain - if I don't have loop set long enough after part I want to record, so I can run back to computer and stop it - so a new loop starts, it will split all takes down the lanes where last not complete recording stopped. I found something in help on Sonar, "delete last take" or similar - that could serve as undoing just that one. Not sure if there is anything like this in current Cakewalk. There were a lot of options in Sonar 4 as I recall. I could design loop recording exactly how I wanted, and where it actually muted each track as it looped. When practicing vocal harmonies and wanted to hear previous take, I could get that - or each take were muted(I got full new tracks then each take). There were a bunch of options to read up on - but I could get whatever functionality I wanted.

Experimented further - set to overwrite then I don't have to hear previous loop as recorded while recording new take, that gave me hope. But when I stopped recording and then start another - all takes that exist in lanes are deleted, take lanes are there, but clips are gone, wiped and gone. So you cannot do 3 loops, and then stop, and then start to do another couple of more loops. Something is probably not as intended here. I just want for practical reason loop record, and make several tries, kind of. And obviously don't want to hear previous loop take as a new is started - I want a hole there as when doing first take in the hole.

Hi Currently have - Comping set - create new lane for take But still get this split action when overlapping. Is it possible to get it just actually mute previous take when a new lane is created - when overlapping in same pos as loop recording do? I mean on the Mute button, not this mute partial clips thingy. So I tried punch in recording(mute takes too), and thought this would fix it since same border each side.. But if you stop in middle, it do this split action on last full take. For me it would feel cleaner if take lanes were left alone until I want to fiddle with it. Obvious half dones I could just remove after so can pick and preview from those left. It's interfering too much. When you do a new time line position on same track it does nicely put the first in lanes up on track. But when partially overlapping even better if left alone. I even started creating new tracks to get rid of this handling, when I have a clip that is ok, but when recording next on timeline you start just a tiny bit overlapping and it's a mess. Is there any settings I'm missing? Thanks.

Missed that, thanks. That's how it goes searching for words and and up in the middle of something. :)

Where did it go? It's not in my 2020.09 but in manual from earlier. Thanks.

I did a walk through how Cakewalk works in the regard when midi CC#7(volume) is sent. Unless you touch a midi track faderin Cakewalk, no CC7 is sent. In Reaper when I ran v3/v4 you could activate that fader is midi - and it is sent every time as you start transport, all 16 channels sent even an initial value as transport is started every time. I had some issues with this how it worked. But if you happend to touch it, it stays there in Cakewalk and the synth is playing back on that volume, which could be very different next time you load the project. So something to watch out for. My link explain Cakewalk a bit, but also that modifier on midi track are after plugin bay for midi - so a midi monitor could not see what is actually sent as modified. For newcomers I link to some decent plugins for monitor, TrackDiag and PortDiag anyway, very useful. PortDiag do realtime through monitoring and playing, TrackDiag what clips on track sends. http://www.tencrazy.com/gadgets/mfx/

Followup on my surround experiment. First - producing a surround mix to a surround bus is excellent and easy in Cakewalk. I don't know how many hours I spent in Cubase a couple of years ago to do the same - just getting meters and ability to send a track as a position in surround. So went on with the idea to produce Dolby 5.1 in a video, to send over spdif to surround phones I have - to see how well it does. These phones use spatial processing with just one driver in each ear - and result is disappointing. Forward and backward position is pretty much from your eyebrows to your butt. There is never the true sensation of an audio source is x meters in front of you, or behind you either. Real closeup positioning works a little bit - but overall I feel this is a joke. So the spatial algos Sony use are not impressive. So consuming many dozen YT demos on binaural 3D sound to phones is not any better either most of them. To fool brain into getting true positioning of an audio source is just not there yet. Anyway, I got the test done like this - creating a 6 channel wav from cakewalk - importing into PowerDirector and produce a video with Dolby 5.1 audio. - setting up MPC HC player in pc to send this audio as spdif output to my phones So in depth some crucial parts were on my RME setup to be fooled sending Dolby 5.1 to a single spdif out - no ADAT outs could be hidden(nothing connected now, but still important) - setting multichannel out in RME as well - no speaker out in rme either - SPDIF out and Analog 1+2 is also there separately Going to windows Sound, - a ADAT rack unit icon appears ADAT 1-8(it did not before having less than 8 unhidden) - setting this rack unit to ADAT 1-6 16 bit 48k Then in MPC HC player - audio render set to ADAT 1-8 rack unit(now containing only 6) - video audio out to SPDIF This to play back on pc. Hardware media players setup for surround audio output will work as well with this video of course. To get really good surround feel I think you need physical speaker setup - so the direction straight in front or behind you is there in a physical way. This as technology is today IMO about creating true surround to fool brain into believing there is an object in any spot in 3D environment. Sideways it works to a degree muffling sound due to obscured angle from a source to each ear - but true in front or behind is a disappoint ment so far for me anyway. But will get there eventually most probably and will be cool when you can create that image on just one driver each ear. So for now I am dropping the idea to make binarual 3D mixes for youtube purposes that anybody could enjoy.

Thank you, thank you, thank you. I always forget about those menu options. Yesterday I looked through preferences and menus for meter settings I was sure was there, until I found it looking in manual on those menus in track and console view as well. It's a bit confusing, I think, not on main menu or preferences. I have to shape up.

Hi Keep being bothered how selection in ruler changes it's range as I select a clip and various things. I remember in StudioOne there were loads of settings regarding this - what kind of selections you wanted - and not. Anything like this in Cakewalk? I basically want to draw myself, or doubleclick somewhere or similar to get selection on clip range or between markers and such. Thanks.

I do mp4 all the time, but it is H264 inside that container. MP4 is not just one thing. And a codec might be missing. It maybe just be sound encoded that is not supported to fail. But if no other solution you can use AnyVideoConverter Free https://www.any-video-converter.com/products/for_video_free/ You can convert to just about anything to anything. I do mp4 from youtube downloads too, works really well.

Just a few things I experienced, having offline computer always as working in daw - software trying to go online for various reasons, like looking for updates I had consistently within 10-12 minutes from start of system this caused crackles. On Cubase Pro so bad that audio engine restarted completely, 7 threads restarting, so if recording then audio engine and recording would stop Locating all software doing this checking for updates, or something towards internet - this went away. Now remaining I had only on every hour of uptime of system left - like 1:00:02, 2:00:02 uptime etc consistently creating the same problem. Cubase Pro stopped recording, and produced serious crackles playback as well. Cubase Elements, Reaper, Sonar, StudioOne only produced a minimum pop that these occasions - no recording issues. I never located this and what produces it. Worked for 3 weeks to solve it, looked for all kind of threads started then, there were none - so obviously in a thread that were running already. If I were online while this hourly, it happends this went away - so something is trying to access internet and unable to. That's how far it went. I used SysInternals ProcessExplorer to open windows for every process, and filming screens to look over if anything happends in a certain process, no sign of activity. Looked through Task Scheduler which had some stuff for hourly test, that I disabled, but no difference. I currently get a small pop in playback in Cakewalk, and just live with this. Nothing affect recording, nor offline rendering - just an hourly pop if transport is running. It's on Dell Vostro 430 with Windows 7 for me and you see uptime in Task Manager. They changed so much in Windows 10, not sure if there anymore. Just a few ideas.... ...one more thing I found in the process - cpu throttling, so look that performance profile or whatever it is called is only 100% minimum and 100% maximum cpu. But usually seen as if latency is below 200 us normally, there are spikes at 1500 or above at the time.

If miking amp does sound muddy and I came to think if what I did at the time I had situation to play amp at normal levels to mike up. Miking position is the key, to get this right I did - play at very low volume and had headphones on, so I get very little bleed right over to phones, you should clearly here through phones at least - try angles and distance and position of mike in front of cab - different distance give different low end more than anything(almost 20 years ago so don't remember details) - angle just a few degrees and change position to cone also is a trial and error - then when getting the amount of low, mid and highs balance - do at normal level and do recording So being muddy in you case, could be too much low end, increase distance to cab with mike. At the time I liked like 10-15" distance and not angled much. Some just put right at cab slightly from center of cone, but I did not like that at all. You should be getting the most of the mike like this. There are various attenuators out there to get volume down and still crunch you want. I got the www.tedweber.com attenuator elements long ago and use Radial JDX DI which is connected between amp and load. JDX also has speaker emulation that I am happy with. So no fuss recording at all, like with mikes.

Thanks guys. I'm not looking for anything other than fixing an anomali in Cakewalk Solo Override compared to any other daws with this solo defeat/safe/save/exempt that exist. It's very useful feature when you can set and forget - and it does not work this way in Cakewalk. There are some strange Solo behavior that appear in some session after an hour or so, where nothing solo works - and restarting Cakewalk fixes it. Suddenly no sound on busses unless soloing them as well etc. But much harder to track down enough to assist Bakers how to reproduce. So just focused on Solo Override for now.

What I did to remedy similar issue - I took the video into PowerDirector(or any video editor) and offset it in timeline 8s, just about bar 5 in Cakewalk default. Then rendering this, it will not re-render video to reduce quality using smart rendering, but inserts black frames at start of timeline before video start. This way trim will work as in your case when imported into Cakewalk. Just adjust to sync with scenes or lipsynk you need. I zoomed in to individual frames, and then set bar 5 exactly on first frame - since I start projects at bar 5 for various reasons.

Thanks, yes I tend to use too many words. I did send an email to support yesterday and reported what I believe is the essential content of it all. - what is happening - what you expect Most obvious is that activating solo override, does not solo anything by itself - on other daws. To test Protools call it solo safe. Cubase call it solo defeat. Reaper call it solo defeat StudioOne call it Solo Save. Very handy to just set any effect busses you create to solo override, you can solo any track and if sends to a effect bus that sounds too. Saves a lot of clicks every time you want to solo something. Time saver for sure. There is some setting link prefader send to mute or similar i aud.ini, and to alter standard behavior of solo override could be done in a similar way, if they consider it as intended as it is now.

Ok, thanks, so I explained badly. Cubase, Reaper and StudioOne. And just trialed Digital Performer, but it was there too. As I understand Logic Pro too, on Mac then(just googled quickly). Logic Pro https://www.logicprohelp.com/forum/viewtopic.php?t=57298 - on the effect busses you have you activate solo override(or whatever they called it) - but it does not solo by itself, it's just there prepared for when you later on want to solo some tracks or busses - so creating a new effect bus you just activate this and forget about it. So when eventually do solo on a track, this track is soloed, and the solo override you made before does not get muted This is what I call set and forget. They are there and does not bother you until you Solo a track or Bus. In Cakewalk as soon as you do Solo override on a bus - it solos this bus???? It has the same effect as doing solo itself - so it serves no purpose. So you have to go through possible effect busses and solo them, as well as any other track you want soloed - every time you want to solo a single track. Then not wanting solo the track anymore, you have to undo the solo on the effect busses too - every time. I mean this is the idea behind solo override(solo safe, solo exempt) - to just set those you don't want muted when soloing one or a couple of other tracks. Like you do for recording, and using DimOnSolo - you solo those you record to stay more in front, but to get effect busses you want non-muted you have to fiddle with that soloing them as well - every time. I see no difference between Solo and SoloOverride as it is now - since SoloOverride is soloing that bus immediately. There are other weirdness with Solo in Cakewalk, and nothing works eventually in a project while working, but closing and opening again fix that. So something about Solo needs some attention in Cakewalk, I think. I am trying to collect reproduce procedure to report later. But some things like regarding if to have sends muted when track is muted has settings in ini-files. You can configure how you want that. So thought maybe something is there regarding this SoloOverride too. But it might be just something broken regarding this SoloOverride or I am missing something.

I keep coming back to solo issues. It' so easy to set up Solo Override in so many other daws, what you know is effect busses, metronome etc and never have to deal with it again. But in Cakewalk Solo Override activates solo on that track. So you always have to set up and then remove with any solo thingy you are about to do. Serves no purpose IMO. You can just use solo if you have to redo every time. All other implementations I found it is just there if you do ordinary solo on any other track too. Thanks. No special settings anywhere for this to work like in most daws?

I'm just intrigued by this new stuff for 3D audio for phones - Windows Sonic and Dolby Atmos. To normal stereo phones you produce full surround in as many channels as you could wish for. But will test now my surround phones from Sony which take 5.1 or 7.1 input and will see how good positioning they do. Front and back is usually what is overlooked what I saw so far. It's just becomes in your head. This newer technology do true spatial processing into normal stereo phones - and that would be true 3D audio. I'm bought this Wavearts Panorama 6 but it does not really do front/back positioning, it's all in your head on left side, over your head and to right side. Problem with both Microsoft Windows Sonic and Dolby Atmos is that their business is built on selling decoders and/or encoders. So you have to write for their API systems to target these decoders they provide for this. I am hoping to find somebody that did both encoding and decoding like this in one box so you could feed and mix surround into true 3D audio into normal stereo phones. I just love this idea to target stereo phones and do all kinds of positioning of music and effects. So many people listen inear phones or phones anyway today. Here is good readup on windows sonic https://www.howtogeek.com/356481/how-windows-sonic-spatial-sound-works/

Wow, you tested - that's incredible, thank you. :) Seems to pickup directly from within Cakewalk then, so outs not that important. But will test both, and feed back here.

Thanks. So output is taken directly from buses inside Cakewalk and not dependent of that ASIO sense these outs as well? And 6 channel wav file can be generated - not as forced on input to mono channels? Can they even be unassigned hardware out if not needing to preview then. I can route this other way while recording and need listening what is there.

Hi As I understand I can use 6 channel wave files. And you can mix to surround bus. If that bus is last before hardware outs - is audio to render picked up at bus out, or does it need hardware outs? I have stereo out on RME HDSP 9632, and a free unused ADAT optical out, and figured I can route 4 outputs to these. But since no DA is connected to those, will rendering 6 channel work? Is ASIO used for rendering so it must be picked up another way to real DA converters? I thought I would make test patterns for 5.1 sound and maybe 7.1 and render in PowerDirector as Dolby or DTS and test surround phones how well they position channels. But would like to user Cakewalk recording the audio. Or if there are such test patterns to download, that could be a cool thing to test as well. Thanks.

Not all sure about Cakewalk, but many daws have a setting whether clicking in empty area would move Now time cursor. Look for something like that.

This happend to me long ago when I had windows sounds active - and running project at 48k and maybe get a "ding" sound from windows when a dialog pops up and similar - could be enough for soundcard to switch sample rate to what windows use. But this was a rather fixed change in pitch then, at the time Sonar could run at 44k suddenly. Just an idea, if any activity is connected to this happening - turn off internal soundcard in bios and also windows sounds.

The reported latency is adjusted automatically, so clips are moved back by that amount, I think even roundtrip latency as calculated is what is used. But as said, that means that interface must have AD and DA encoders on them, so that is reported too. Pure ADAT cards like RME RayDat does not have encoders, so extra values for AD and DA converters delay must be added. My RME HDSP 9632 has if it was 149 samples on DA and 111 on AD. And strangely enough the piggyback addon card AI4S is different by 6 samples. My Audient ASP800 over ADAT is 3 samples quicker than built in. These things are easy to test out how clips are placed re-recording like metronome sound or something else on timeline. Just do loopback with cable. But just for knowing I did this, I don't bother adjusting by nudging all these depending on inputs unless multimike something on different delays. I did both line input and through other gear where inputs were on different delays, so adding these later it sounded boxy already, just having different by 6 samples. I think a delay field for each audio track would be great addition too Cakewalk. I have this noted, as well as how long response delay there is for external midi gear to set negative ticks delay on midi track to make that align as rendered. Then if plugins introduce delays that can be turned off with toolbar button [pdc] if you don't want that delay added as tracks are played back. But there is usually no need to have everything playing back doing recording, so either minimize what is playing by disable tracks, or disable plugins is also alternative or turn off pdc.