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Noel Borthwick

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Everything posted by Noel Borthwick

  1. Yep we can see what every user is doing onscreen now in real time 😛
  2. WASAPI should be available on all Windows installations. Its the default Windows driver model. Click on Playback and recording and then choose "Wasapi Shared" as the driver mode. That should work with Realtek sound devices. >>As well as the observed symptoms, using ASIO mode against the onboard Realtek HD Audio causes hundreds, then thousands of threads to accumulate in the Cakewalk process. Wow thats bizarre sounds like a serious driver bug. The threads cannot be from Cakewalk and are likely being spawned by the driver itself. Unfortunately we don't have the Realtek ASIO driver since I think it only ships with some OEM PC's.
  3. For DX plugins it should automatically load the 64 bit version assuming it is installed and working properly.
  4. It is possible that you may see an improvement when switching lenses since there are some optimizations in the load code that prevent track settings from being updated among other things. I don't follow how you imagine workspaces would assist in the other thread. Workspaces essentially act like a filter for visible features and store positioning information for views and toolbars. It won't allow you to move where a feature itself is visible.
  5. @Colin NichollsNicholls can you post your report here please? Early access is better handled here rather than the support channel.
  6. 100 ms isn't that much for bounce. Plugins should be able to tolerate any buffer size. I suspect one of the plugin's in the project has a bug with large buffer sizes and is corrupting memory when its set high. Since your audio interface won't go higher than 20 msec you can try switching to WASAPI exclusive or MME. Both those modes allow the audio buffer size to be set as high as 200 msec. If the export crashes with those settings then we know that its related to some plugin in the project..
  7. @Joe Dun thanks for your feedback. Simplifying the program while retaining behavior for professional use can be a complex endeavour to do properly. We are committed to improving in this area and the Lens feature was a step in this direction. In the next update we are making this much more discoverable and have added it to the first run onboarding to allow users to pick a preset that best matches their requirements. In the future we plan on going even further with this so please keep the suggestions coming!
  8. Interesting. Thanks for troubleshooting, @petemus. All that setting does is set the buffer size to that prior to the bounce and then restore it post bounce. Can you try and set your audio device to the same latency if it goes that high and retry the bounce? This should do essentially the same thing so it will be interesting to know if the problem recurs.
  9. If you are getting heap corruption it's likely one of your plugins corrupting memory. So not something we can solve necessarily. Try and isolate which plugin causes it be removing plugins one by one and exporting after each step.
  10. Yes Lenses have incredible power to customize the app to make it as simple or complex as you want. As you discovered you can completely remove a feature you may rarely need such as screensets by making a lens that excludes it. The same could apply to any other feature, or views. The big deal is that not only does it hide the UI for the feature, it removes all the hotkeys associated with it so that you won't accidentally invoke that feature. For users who want a customized de-cluttered experience that is focused on their needs lenses are an ideal solution. For the next release we have made several improvements to lenses. We're also renaming the feature to "Workspaces" to make it more evident to new users what this feature is intended to do. There will be some improvements to the factory lenses, several fixes and also an enhanced onboarding experience for new users to initiate them into using the feature.
  11. 99% of plugins handle it with no problems. I never use anything other than fast bounce.
  12. Are you using fast bounce? That mode never drops out. If the bounce fails you should get an error message.
  13. @chap_sistine where are you getting a blank screen? In BA or CbB? BTW The new BandLab assistant 3.1.1 has a mechanism to manually refresh activation of CbB without having to reinstall the application or log out and back. See this screenshot. To access that option click the dropdown menu next to the open button.
  14. Please send the CWP file (for the case where it doesnt work) and we can look at why you don't see pan.
  15. Hangs and crashes are two different things. It looks like you are seeing a hang. If you capture a dump and send a link we can investigate and advise what is causing the hang potentially.
  16. If all you have is camera audio and external audio that you want to sync together its pretty simple and doesn't require any special automation tools. I do this frequently to match audio recorded from a camera with audio from a more high res source from live recordings. All you typically need to do is this: 1. Import your camera audio into cakewalk. File | Import | Audio and choose the Video file itself if you don't have the audio extracted already. Cakewalk will rip the audio directly from the video assuming the Video format is compatible. If not you will have to use an external program to rip the audio. 2. Import the external audio into Cakewalk on a new track 3. Maximize the two tracks and zoom in so that you can see the waveforms in detail. 4. Drag the second clip to align the waveform start. If the waves don't start at the same time you will first have to manually locate the start of the audio in the recorded track by listening. 5. Zoom in as far as you need to make sure you perfectly line up the start of the audio. Even in live recordings you can find a transient peak and line up the audio accurately. Use your ears to test. Once its aligned you should be able to play both tracks and not hear any significant flamming or phasing. Note that for live recordings you won't get perfect phase alignment. Its not something I worry about too much since you don't typically run into phase cancellation with such sources. In most cases this is all you need to do. In rare cases if the clocks on the devices are very different or bad you may get some drift if its a very long audio file. I personally have never encountered this with any current generation cameras. Even with audio lasting an hour or longer the drift is negligible. If you have drift, locate the areas where you visibly see the waveforms diverge and split the clips at that point. Now you can manually align by repeating the process above. You can also time stretch one of the clips to match but if the drift is non linear the manual method is the only way. To time stretch a clip, hold down CTRL and SHIFT simultaneously and drag one of the ends of the clip. If the audio is just dialog you can use our built in VocalSync, which is designed primarily for the purpose of aligning two different vocal clips.
  17. No objections to sending dumps. developers find them useful. It may show them what is taking long if you capture it with a project only containing ozone.
  18. Its not impossible that its a bug in the state of the input monitor channel in CbB. Do you have "Allow arm changes during playback/record" enabled? If also see it with another interface please let us know. Its strange that you see input on the layla panel. However its possible that its not passing the input to the DAW via ASIO.
  19. I used a basic Ozone preset "Clear Transients" that was light in processing. Windows version 1809. I'm running the older focusrite driver (focusrite control 2.3.4) because the new one bluescreens! Prior to this I don't think 16 samples would work properly since there was a limitation of 1 msec minimum being enforced internally.
  20. If you are seeing big differences in load times between product versions it's best to contact izotope directly and let them know.
  21. Holding down shift engaged the quick grouping feature which made it extract timing from all selected clips. If you want to extract timing from a single clip don't hold down the shift key.
  22. @scook I suppose I could drop that size to 32 samples. The thing is that the smaller the buffer size is the higher the CPU load is, so if someone is turning on load balancing to primarily save CPU this could not be useful. Give it a try with 32 or 16 samples and check if you see benefits when running your audio interface at 48 or 64 samples. (Load balancing only kicks in if the actual buffer size is a multiple of MinPlubnginLoadBalancingBufferSamples) I'll also give it another look next week. I can't think of any other settings that would be candidates for changing.
  23. The main factor is the driver yes but the net effect depends on both the driver and the host. The host has to efficiently (and correctly) handle very low buffer sizes such as 16 samples to facilitate that.
  24. Right although from the host perspective the RTL isn't something we have control over. I'm talking about the audio buffer size here not RTL. As long as the host and the driver itself can handle very low buffer sizes efficiently (without dropping out) it means that you will get low latency performance with drivers that support it. SONAR had incorrect behavior with buffer sizes lower than 1 msec (< 44 samples at 44100 Khz). By fixing that we have allowed for devices with buffer sizes smaller than that to be handled properly.
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