Noel Borthwick

Please send the CWP file (for the case where it doesnt work) and we can look at why you don't see pan.

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.

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.

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.

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.

I think you may need to break up with Layla

If you are seeing big differences in load times between product versions it's best to contact izotope directly and let them know.

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.

@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.

Do you have plugins in that track? If the rest of the project is playing ok toggling the engine shouldn't have any affect on input echo.

Is it possible that the audio engine dropped out? What that happens is the engine light on?

I wouldn't track a big project at 16 samples. It would be ahem a bit fragile There are also improvements to MIDI buffer performance in the engine coming for the next release as well as an overhaul to ASIO buffer management. I guess nobody noticed a subtle giveaway of some changes in the video lol.

That's the focusrite 6i6 audio interface not a plugin.

We've been working hard on improving the audio engine performance over the last year. Low latency ASIO performance is greatly improved compared to SONAR. Please check out this post to see it in action.

I asked for a video because your problem is unusual and a video will frequently show us visual clues that you missed describing. There is nothing relevant I see immediately that prevents the transport button from highlighting. If you press spacebar instead of clicking the transport does the project play? Also check if the playback timing master is selected in the audio driver settings just in case. If you create a fresh project does the same problem occur?

Sure ask them to contact me. I was previously in contact with Ryan Swannick there regarding a different issue so they have my contact info. The issue is that the IK plugin seems to not like the effMainsChanged and effStopProcess calls from being made early while its being scanned. Our scanner has always done this so its not new behavior. I've removed this since they are potentially unnecessary. We're still testing it in beta to ensure there are no regressions with other plugins.

We have a workaround for this. I sent @ceez an updated scanner that prevents that problem. Its actually a plugin issue however.

Bus names are not important. I see that you have some buses and tracks soloed. If you unsolo everything does it work?

@ceez can you reproduce that scan issue with B3-X? I'd like to send you a newer version of the scanner that work's around that issue to see if it solves it for you.

@Jay MarshallDid you have a plugin that forces mono on the matter bus? If you have a copy of the problem project please share a link and I can take a look as to why that might happen.

If you post a video reproducing the issue, it might help troubleshoot.

Can you post some steps please?

Most likely your issue is that your tracks were going to a master bus which has its interleave switched to mono. If you do that track pans will no longer work until you switch the bus out of mono interleave. The fact that it got solved by creating a new master attests to this.

Without knowing your signal flow its hard to tell but I bet your issue is electrical grounding. For a long time I had a lot of issues with noise from my guitar processor going into my audio interface and had a hard time isolating the source. Depending on your house and quality of the power lines it can be hard to eliminate. Dimmers on the circuits can also cause noise to get into the power lines. In my case I got rid of it by using ferrite cores on all my cables and also using this hum eliminator device. Some PC's are notoriously bad at letting electrical noise get through into your audio interface from. Also ensure you have a common ground for all your plugged in devices.

If you are planning on deleting data from tracks then mix scenes won't save that info so you would have to first save an independent version of the radio edit project. However once you open the radio edit project file you can certainly create and use mix scenes to quickly switch between your original mix and variations of the radio edit mix. You will get the best of both worlds that way.