Bass Guitar

Wow. I'm surprised anyone would answer this after they were rude enough to not answer the first time for a year and 2 months.

That screenshot of Latency Monitor looks pretty good to me. Latency monitor always seems to point at my graphic drivers no matter which computer I've run it on. I actually just ran it yesterday for the first time in a while and I was surprised to see red. It pointed at a Realtek audio driver. Of course it's disabled but I thought I had un installed it too? Turns out Windows will re install it again. SO I uninstalled it again. But that didn't get rid of the red? ANd now is showed once again that it was my graphic card ( NVIDA ge Force RTX 3060 ) I can't find the control panel anywhere?? So I will be downloading and re installing the driver after I post this. Then I found that Windows update had also turned on the Wi Fi. That was the issue. I had to go look at a bunch of tweaks I had made and re set them. Same old story with using a computer for audio.

You should read this- Seems having ARM supported CPU will be the future in DAW's . Yes is you do video editing then make sure to spend some cash on a good card. Makes a huge difference when rendering.

Yes works fine for me too in Sonar. I have the version that came with Sonar X3. I most certainly wouldn't try and do much using the trial version it is crippled in too many ways.

@Mauro Gaspa Instead of using a Cell phone which makes the video blurry use this--it's free and used by millions of people around the world. OBS studio. https://obsproject.com/download Having two monitors makes it easy to capture Sonar on the main screen. Put OBS studio on the second monitor. It even allows you to capture the audio. That way you can explain what is happening. And don't worry about using italian language. It's pretty easy to transcribe audio these days. It's not hard to learn how to use it. If you need help send me a PM.

All midi settings like level. Pan, Channel.Bank and program change are handled by the track header. The minute you touch the Pan or the Level that embeds a controller message in the midi data for that track. It will reset on playback. Take note that it does not show on the events list ( ??) Example if you set the pan hard left and choose the Bass patch on channel 2 in the track header, Then in the TTS-1 in Channel 2 choose a guitar and set the pan hard right it will all change back the minute you hit the spacebar.

I just checked and I don't think you posted which Audio interface you are using. This is another factor people have mentioned before. Are the drivers up to date and supported by your operating system. Are you setting Windows at the same sample rate as the Audio interface and Cakewalk? Also Have you tried running a project using only a Cakewalk instrument like SI piano or drums?

Most VST synths don't use banks. They have their own list of presets you access in the user interface. Cakewalks Banks and patches are mostly for hardware and GM players. The only synth I have that uses Patches is Airs xpand2. Example Dim pro has a million presets but they don't respond to anything that is in the Track header settings. I'm not sure about Arturia I used to have it but it crashed Cakewalk all the time so I deleted it.

It's true that S1 and Mixcraft both have a rent to own option. You keep it forever after a year. But right now Sonar is part of a larger package so that's totally a different animal. I see this is what a lot of people are objecting too. But then- Take note that at the sale price of $100 per year Bandlabs membership is by far a better deal than both S1 ( $262) and MIxcraft ( $180 ) Pro Tools Studio ( $311) and Ultimate ( $631) Even at full price Bandlabs membership is a better deal. And personally getting 2 Daws for the price of one is a bonus. Next is a cool little Daw that I find myself using a lot. I just recorded my 45 minute live set none stop. Added a few effects and I'm done. I wish it had better meters. But I had Melda. My only real concern is Sonars future. Will it still be around in 5 years? Everybody has ether changed Daw's or is still using Cakewalk.

What I have found is you need to always make sure your interface is connected and active in Windows before you open Sonar. Otherwise that project will forget everything you set up. When I do multitrack live recording I set all my inputs up at home and save the empty projects. This normally involves 14 audio tracks and 1 midi track. As long as I have the interface running before I open Sonar it always has all the correct inputs set up.

I found this video about getting the waveform back as well. https://www.youtube.com/watch?v=iw_Ff5b4_9E&t=29s For posting videos as said what I do is put them on You Tube but make them private. It's to bad we are each using a different editor but I think they all do the same stuff. Here's a screenshot of a 3 camera performance. You can see how I lined up the count is and then the studio recording at the bottom

Well they will need to get it working correctly in Next before they screw up Sonar. What can happen to me is it will change the project sample rate but the tracks that were at 48 are now totally out of sync. I just tested this. I dropped a finished song into a project that was set to 4800. I used the stem separation which come back at 44.1 and the project changed to 44.1? now the original is out of sync. I looked in the audio folder and the original shows as 48/24 and the stems are 44.1 /16 I have had out of sync issues with a few other projects and now I think I see why after reading that part of the Manual. Its that Auto sample rate that can mess stuff up.

Yes exactly. Up till now all my videos were shot at gigs where I was using my backing tracks. Then all I need to do is drag the backing track into Vegas and sync up the very obvious count in. If I only have one camera this is supper easy. Three camera's took a lot more work but the audio stays synced up no problem. As I said I also have always recorded the vocals and the guitar into their own audio tracks using first a Tascam DR 40 and now it's built right in to my Zoom L8 mixer. Then I also sync those up with the video. Only trouble is you can't hear the count in very good. Only the leakage into the mikes. ( Guitar amp is miked) Anyhow I have lots of options once the video is shot. I have never used a Daw for video work. But This will be my new project where I will be adding real instruments after the fact instead of using backing tracks. No click tracks.

@mettelus which video editor are you using? I'm still on Vegas pro because it sort of works more like a Daw for me. You mentioned separating the Camera audio track from the video track. In Vegas I do this and place them at the very bottom so all the audio tracks are stacked together. This allows me to line them all up best as I can. I find that because video works in a pre frame format I can't always get this perfect. But only my polished up Daw track will be heard anyways. It seems I cannot delete the original camera audio tracks. It deletes the video as well. So I just shrink and mute them. And yes you can actually record audio, and use all your plug ins in Vegas. It will even tool copy the audio into Sound Forge or Wave Lab for fine tuning. I like that it supports DX as my go to compressor is the Sonitus. I then use the Loud Max Brick wall limiter set at -1.0 db on the master track. I can easily dial in a perfect -14 LUFS at -1.0 peak. Some day I will learn how to use Resolve.

Doubt that could be possible. This function comes with a warning in the user manual. Next has to convert the sample rates in real time. This involves your CPU working harder. Example, I work at 48/24. So Next is also set to auto detect my Audio interfaces sample rate. Therefore I record all new audio at 48/24. Once there is audio in the project you cannot change the sample rate to say 44.1 it will pop right back to 48 ( I just tried this to confirm) But you can drag and drop 44.1 wave or MP3's into the 48 project. But the project is still 48. I learned this using the stem separation feature. It brings back the 48/24 file as a 44.1/16 stems. I noticed this and was wondering WTF? So I found it in the user guide. Here's the section in the Next Users guide which is actually very well done: What should my sample rate and bit depth be for recording? For music production, 44.1kHz/24-bit or 48kHz/24-bit is a good choice and provides a good balance between sound quality, file size and processing power, and captures the entire audible frequency spectrum accurately. Although a high sample rate along with a high bit depth will deliver the best audio quality, it is not always the best option. The higher the sample rate and bit depth, the more audio processing power and disk space are required. This will reduce the potential track count and number of real-time plugins, and may not make any noticeable sonic difference. 16-bit audio has a low noise floor and a decent amount of dynamic range, and is fine for most music production. 24-bit provides more dynamic range, and lets you record at more conservative levels to avoid exceeding the level of clipping (0 dBFS) while staying above the noise floor. The added headroom may also be beneficial when editing and mixing. With 16-bit and 24-bit recording, it’s important to set proper levels prior to recording in order to make sure the audio isn't peaking and distorting, while also staying above the noise floor. If the recording level is set too low, raising the track's volume will also increase all of the corresponding noise in the recording. 32-bit recording is generally reserved for ultra-high-dynamic-range recording, and is not meant for distribution. The main benefit of 32-bit float recording is improved flexibility in making level adjustments after recording is complete. There is so much headroom that it doesn't really matter much where levels are set while recording digitally. Any signal that clipped in the recording can be recovered later by reducing track levels. Signals that exceed 0 dBFS can be reduced to below 0 dBFS, undistorted, and any parts of the recording that had low gain can be raised with less chance of increasing the noise floor. This is mostly useful in an uncontrolled environment where you are not able to set proper recording levels prior to recording, such as live recordings, but is less critical when you are able to set proper recording levels ahead of time. Project sample rate. The project sample rate is set in the Project Info Editor. The default project sample is Auto, which adapts to the audio device sample rate. It is strongly recommended that you use the same sample rate for the project and audio device. Using different sample rates will cause real-time audio resampling, which may consume extra CPU. When you create a new project, if you do not want to use the default audio device sample rate, you must choose a sample rate in the Project Info Editor before you start recording audio. For details, see “Project Information” on page 26. The project sample rate is used when recording and importing audio files. The project sample rate becomes fixed as soon as any audio clip is present in the project. When you record or import the first audio clip, the project sample rate will be switched from Auto to whatever the current audio device sample rate was at that time. From then on the project sample rate is fixed. Removing all audio clips from the project resets the project to Auto which means it will follow the device sample rate again. The project sample rate can be changed at any time from the Project Info Editor. Cakewalk Next® will attempt to auto switch the device sample rate to the project sample rate when possible. Auto switch occurs both when changing the project sample rate as well as when loading a project Match the project sample rate and audio device sample rate It is strongly recommended that you use the same sample rate for the project and audio device. Using different sample rates will cause real-time audio resampling, which may consume extra CPU. Real-time sample rate conversion ensures that projects can play back on virtually any audio device at the correct playback speed, even if the audio device doesn’t support the current sample rate. The default project sample is Auto, which adapts to the audio device sample rate. If there is a mismatch, Next will attempt to automatically switch the device sample rate to match the project sample rate when possible. Auto switch occurs both when changing the project sample rate and when loading a project. The sample rate pickers in the Project Info Editor and Preferences > Audio indicate if there is a mismatch between the audio device sample rate and project sample rate. A notification also appears if there is a mismatch.