msmcleod

I use a 23" 1920x1080 monitor in my office (well 3 of them, but Cakewalk is only ever on one of them), and that's fine for me with my reading glasses on. I'm about 1 meter away. In the studio I use a single 27" 1920x1080 monitor at just under 1.5 meters away, which is perfect. I still have to wear my reading glasses though, as anything less than 2 meters away is a struggle without them. I did try a 32" TV at one point, but it just felt way too big.

I guess there's no technical reason why this couldn't be done, but it'd require a non-trivial amount of development work. The biggest issue would be the UI redesign... and those dropdown lists take AGES to populate, so it would take twice as long to open up the Mackie Control dialog. This IMO would override any benefit from having the extra function mappings. In saying that, there's nothing stopping someone doing this themselves - the code is on github: https://github.com/Cakewalk/Cakewalk-Control-Surface-SDK

Ah... the MT-32... I released an album back in '94 where over half the sounds were from an MT-32. The other half were from a CM-32P (the desktop version of the U110). I ended up writing my own MT-32 patch editor for the Amiga 500. I've sampled most of the MT-32 sounds I used, and still use some of them today.

I used to think that, until I realised that the only reason more modern sample-based synths sound good is because of the FX units built into them, and in the case of the soundfonts, putting them through the E-MU filters did wonders. Try listening to the raw samples of a JV1080 and you'll get what I mean. Likewise, any early to mid 80's digital synth can sound amazing just putting it through some chorus & reverb, and even better though a decent multi-fx.

It's a fair comment. Sound travels at 1.13 feet per millisecond. 21 samples @ 44.1K is 0.476 milliseconds. So it's the equivalent of moving a mic approx 6 inches further from your guitar cab, or you moving your head 6 inches further away from your monitors to listen to them.

Take a look the following settings in Preferences, and see if changing them makes a difference: From the help: "Suspend Audio Engine When Cakewalk Is Not in Focus. This option allows other software to access device drivers. When this option is checked, other software can access device drivers when Cakewalk is not the focus of Windows." - If this is set to true, Cakewalk will be closing/re-opening the audio drivers every time it loses/gains focus. "Always Stream Audio Through FX. If any tracks or buses contain active plug-in effects, the audio engine will be activated and stream silence through the effects, even if the tracks have no audio data or Input Echo enabled. You will typically only disable this option if you want to conserve a bit of CPU processing if you play a project that contains lots of empty audio tracks with effects. There are several benefits to having this option enabled: Any effect with a “tail” (reverb, delay, etc.) will finish playing when playback is stopped. Effects will respond to automation envelopes, even on empty audio tracks. Many plug-ins do not update their UI properly until they receive audio input." - If this is NOT checked, then the audio engine will be disabled when you stop playback. This will create a delay when starting playback again - the length of this delay is also affected by the option below: "Always Open All Devices. With this option checked, Cakewalk opens all enabled stereo pairs of audio outputs as soon as you press play or turn on the audio engine. Any pairs that don’t have any tracks feeding them stream silence, but are still ready for use. So if you then change a track's output assignment on the fly Cakewalk doesn’t have to reopen a device, which can cause gaps in playback." - If this is checked, all device channels will be opened at once, so if you have a large number of them this could take time. As long as "Suspend Audio Engine When Cakewalk Is Not In Focus" is unchecked, and "Always Stream Audio Through FX" is checked, this shouldn't be a problem, as it's only done once. But if this isn't the case, then it could take more time to start playback. In most cases, it's preferable that this option is enabled, and you simply disable the channels you're not using.

I can't seem to replicate this behaviour - when committing an arrangement, I end up with exactly the same amount of take lanes as I started with. It may be project specific, or perhaps some setting is causing the issue for you. Can you PM me your Cakewalk.ini file ( it's in %APPDATA%\Cakewalk\Cakewalk Core\ ), and a registry export of: Computer\HKEY_CURRENT_USER\SOFTWARE\Cakewalk Music Software\Cakewalk\Core That will allow me to replicate your current settings and hopefully reproduce the issue. If I can pin-point exactly which setting is causing this, I'll let you know. Also, can you clarify - is this audio clips that end up in new take lanes, MIDI clips, or both?

It could be that the stretch algorithm you're using isn't suitable for the material. Also, note that there can be different algorithms for "online" and "offline" stretch. Changing the tempo value in the inspector will use the "online" algorithm, which may be fast but not as good quality. Selecting the clip and selecting "Bounce to clip(s)" will commit the changes using the higher quality "offline" algorithm. Alternatively, you can set the online & offline algorithms to the same higher quality algorithm. Stretching will be slower, but will sound better and you won't need to bounce to clip(s) afterwards. You can set the algorithms in preferences:

In WASAPI Exclusive and WDM mode, you should be able to use more than one audio interface at once, however as @scook has indicated, unless they have their word-clocks synced, you're likely to have some timing issues. If you're just recording from more than one interface, this may not be too much of an issue... If you set up your channels so that each instrument goes to its own separate input, but one instrument goes to an input on both interfaces you should be able to compare the two tracks of that instrument to see how much they're out. So, for example... you've got your Focusrite set up as both recording and playback timing master. You record all of your instruments together, with say the kick drum, going to an input in the Focusrite AND an input in the M-Audio interface. When recording stops, compare the kick drum track from the Focusrite and the M-Audio interface - you'll need to zoom in enough to see the waveforms. Now select all of the tracks that were recorded on the M-Audio interface, and while looking at both of the kick drum tracks, drag all of the selected tracks together so the waveforms in both of the kick drum tracks align exactly. Now they should all be in sync. You could actually take note of the sample positions of a known peak in each kick drum track before you align them, then subtract one from the other, so you know that's the amount you need to adjust all of your M-Audio tracks by after recording. You could even set one of your Nudge amounts in preferences to this value, so you can re-use it in the future for adjusting the tracks recorded on your M-Audio. I'm not 100% sure this value will always be correct, but it should be good enough within the same session. When playing back, make sure you only playback through your Focusrite.

Unfortunately it's not easy to solve. but it is high on our to-do list.

Anything that lives in the multidock can be undocked and moved to another monitor.

ooh... this sounds like a 3rd party plugin that isn't authorized. This can happen occasionally with some plugins after a Windows update. You'll need to bypass all the plugins in the offending project and enable them one by one to find out which one is the culprit.

I use a couple of these with Cakewalk occasionally....

This is almost certainly nothing to do with Cakewalk, but more likely to do with a mismatch in sample rate between Cakewalk and Windows. If you've got Cakewalk is using the same audio interface as Windows is, make sure you change your sample rate and bit depth in Windows to match what is set in your Cakewalk project. If this doesn't fix it, I recommend re-installing your audio drivers.

If you've got the space, don't discount a mixing desk with direct outs. I'm using an Allen & Heath MixWizard WZ3 for 16 inputs, which has a HPF @ 80Hz, hi/low shelving and two semi-parametric mids on each channel. The WZ4 is the updated version which retails around $1800 new, but you should be able to pick up a used WZ3 for around $500. The pre's / EQ are incredibly clean on this board, don't go for this if you want a "character" eq. For another 8 inputs, I've got an old Alice 828 with Hi/Low shelving and a parametric mid, although I mainly use this for the saturation given by the Belclere transformers. The new Mk3 will set you back around $2500 (the Mk3 has Carnhill transformers as they own Belclere now, and only 3 mono channels with EQ - the rest are stereo), but if you can find an old Mk1 or Mk2 with the full 8 channels, they go for around $1000 used. Another 6 inputs use the TFPro P3 (basically a rebadged JoeMeek MQ3), which give me an optical compressor, and 3 band EQ on each channel. These go used for around $150 - $200.

There are two ways of dealing with SysEx data, either in Banks or as inline events. In Preferences->MIDI->Playback and Recording, there's a "Always Use Sysx Bands for MIDI Files (Instead of Sysx Data)" setting, which I believe is checked by default. If this is checked, then any Sysex data will be moved to banks - if it's unchecked, it'll stay inline in the sequence. Banks can be easier to deal with in most cases. The only time I ever use inline events is when sending firmware updates to hardware over MIDI.

As an alternative, you could get another device that sends out sustain messages: https://www.thomann.de/gb/midi_solutions_footswitch_to_midi_controller.htm The advantage of this device is that it merges the MIDI output of your piano with the pedal CC64 messages, so they come out of a single MIDI output. It's pricey though. However, a much cheaper solution (which I used myself for a while) is to get a Miditech i2-mini 32 (£36), and plug any generic sustain pedal into it. You can get sustain pedals for around £10 - £20. Stagg do one for £10 : https://www.amazon.co.uk/Stagg-21203-Keyboard-Piano-Sustain/dp/B00EZPBA0U/ref=sr_1_2

Cakewalk's clip stretching algorithm preserves the pitch, so it's unsuitable for this task... however, if you've got an older version of SONAR installed, you may have the "Time/Pitch Stretch 2" DX plugin, which should do the job. e.g. these settings will play it back at half speed and pitch: However, any audio editor should be able to do this, e.g. SoundForge, Acon Digital Acoustica, Audacity etc. Audacity is free: https://www.audacityteam.org/download/ You can get a free version of Acoustica Basic Edition 6 here: https://acondigital.com/software/AcousticaBasic6_0_19us.exe Drag the clip from Cakewalk to your desktop (or any folder), then open the file in the audio editor, do your processing, and import it back into Cakewalk. There are ways to integrate the tools into Cakewalk's Utility menu so you can run it directly from Cakewalk, but this is a bit more advanced, so probably best to keep it simple for the moment.

Although it may seem that the hierarchy is Tracks->Lanes->Clips, this isn't actually the case under the hood: Clips are directly owned by the tracks. The lane that a clip appears in is purely a UI concept, with the only exception being lane solo/mute, where some special processing is done to exclude those clips from the audio graph. Adding an effects bin to a lane would not be an easy task, and as others have said, it's something that can be easily accomplished by separate tracks.

I've managed to reproduce the issue when the clip is close to zero, and you move 10ms right then attempt to move 10ms left. It's now fixed for the next release. The issue was that because the clip is in musical time, but the nudge is in absolute time, it has to temporarily convert the clip start time to seconds to check it can move to the left. However, the start time is coming back as a clip start of 0.0099999999 seconds rather than 0.010 seconds. As it thinks this would result in moving to a time less than zero, it abandons the operation.

1. What are your "Nudge 1" settings in preferences? 2. What is your audio clip's time base? If you open the clip inspector, and look at the start time, it should move by the same, but opposite amount when using 1 & 3 on the keypad. As long as it's doing this, there's nothing wrong with nudge. Also, bear in mind that if you've got "Nudge 1" set to follow snap settings, then landmarks could also come into play. This might explain why they're different.

It's a recent one - I got the email about 5 hours ago.

@Andres Medina - thanks for the report. It's been fixed for the next release.

Check your timing master settings are set correctly: And check your clock source for MIDI: The clock source for MIDI should always be audio, unless you've only got MIDI tracks and all of them are routed to hardware MIDI devices.

It Bites - Kiss Like Judas: