Bass Guitar

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.

I believe you need to be using professional cameras that can record a syncing time code. Using podcasting cameras or a cell phone you're going to have to use the old fashion method of having the Clap board. Myself I make sure all cameras are recording from the start to finish. And that all cameras are recording the audio. I make sure there's some sort of a count in and I line that up at the start and it's usually good to go. It's super hard to add video footage that starts later on the time line. I have two methods I might use. At a live show- I video record myself singing and playing guitar along with the pre recorded backing tracks. I get someone to run the camera and a few times I set up 3 cameras. My mixer is a Zoom L8 so I can record the show to multi track audio to an SD card directly. I have good audio quality but sometimes I suck and it's not that great. So I cheat and back in the studio if the live vocal sucks I just use my studio recording and avoid close ups of my lips! I fade in the intro and the applause at the end. If the vocals are good I will still add the studio recording of the backing track. For sure I try and keep the live guitar solos. For studio video recording I'm still working this out. But my first attempts are recording my myself as I sing and play guitar as well as I video myself playing the other instruments. I just got a Zoom Q2n to do this with. As the camera records the audio is being recorded to my Daw and the only hick up right now is I sort of have to do the whole song without screwing up when shooting the first part with me singing. The other parts are easy to cheat. Otherwise I would have the problem you are having! It's either that or get good at lip syncing. That ain't gunna happen. Note that this project I'm not using backing tracks but a snare drums and Bass. Then I might add a lead guitar. If I use backing tracks I would then probably use my in ear monitors.

Ya I see what you are saying, But that doesn't really help much. No matter what you try and name them all inputs are paired together. And the default for inserting a new track will be stereo even though in my 20 years of recording I've never used stereo. And even though you know this every once in a while I accidentally end up recording a stereo track. It's just my observation that Sonar and Mixcraft are the only Daw's that show your interface this way. And have no setting for a default choice of input. Thuss you use templates.

Sorry I guess I should have looked it up first. I've never run into this before. I guess you're stuck with using 96. Not really a big problem considering how big and cheap storage is these days. Like the guy yesterday on another forum when asked why he was mixing down all his songs to MP3 said to save space?? His question was "why do my songs not sound as good as professional releases?"

OK so you say you can rename inputs? where. I looked through all the preferences tabs.And in the case of the Zoom the Master gets called input 1/2 which is what Zoom chose to do. You normally never use the Master, As a result Channel 1 is called input 3. In Sonar if you are new to all this this list is totally confusing as all the even numbers are missing. ( Note that only Mixcraft has this same system) My best example of a very good preference audio input set up would be Studio one. Rename, re order, save the set up for later recall when you change interfaces. And each input is listed separately as Input 1, input 2 etc. It also remembers all my interfaces. It allows you to insert audio tracks that are sequenced from input 1 through all available. It even gives them different colours. Cubase not as good but still each input is on its own. And Waveform same thing as well as allowing sequential inserting of multiple tracks.

Well the Op made a bit of a bad choice by using a high clock rate just to reduce latency. They could have simply have used direct monitoring. So they at least have choices. And as I said you use a Copy. If for some reason you run into the situation that requires a return to the un compiled tracks that's easy to do.

That's funny that staff don't know how to do this. It is actually very simple. I do this in reverse from 44.1 to 48 but it should be the same. Best to work with a copy of the project As said midi data is fine. Leave it alone. All audio track will need to be exported as stems using "Tracks no automation of effects" Export as 48/32 no dither Select that they all export at 1.00:00 Entire project you need this so it's easy to line them up later. Open all automation lanes( you don't want to delete this) Now delete all audio. There cannot be any audio left in the project. Check in the track manager for hidden tracks. Once all audio is gone click in the Transport on the 96/24 to open preferences/ Driver settings. Open the ASIO panel and change the sample rate. You will hear a pop or click and the transport should now show 48/24 Now simply open the browser locate the project export folder drag the audio back to the original tracks and re save the project. Done it dozens of time and now all my old projects are 48.

Looking at different Daw's I found Sonar has the most confusing input list of all with Mixcraft a close second. I'd score Sonar at almost the bottom if you were doing a shoot out of input list features. This is especially nasty when you use an interface with a lot of inputs like a digital mixer. Studio one as example allows you to totally customize the list, rename, re order, mono, stereo etc. You can then save this set up. Very handy when you switch between interfaces a lot.

There's no point trying to communicate with staff here as they don't have time to read most of this stuff. If he PM'd you that is how you now communicate with him. They will sort out you problem even though technically it's not their problem. That company is long gone. There just nice guys. You should forget that POS X1 anyhow, possibly the most bug infested version of Sonar ever! Just install CbB Last time I checked it still runs on W7. At least I have it on an old laptop here somewhere.

Definitely something not right with graphics. Try this. Dump a movie file into Sonar. Now open the video view and undock it. Play the movie and stop. Now resize the video view. It will not refresh to the new size until you start playback. Switch to the Console view then back to the video view, Still shows the console view until you start playback. I don't have Cakewalk but I'd be interested to see if it behaves the same way because I sure don't remember this. I have a few other Daws and none of there Video preview screens behave this way. It's annoying and makes Sonar unusable because you can't stop and edit the audio when your trying to see when the drummer hits the snare! I'm seeing weird performance issues like this as well. Hard to put my finger on it.

Exactly! It was a band aid solution in the day's when a Sound Blaster was one of the few audio cards sold in local stores. It has served it's purpose up until Windows updated the audio drivers to the new WASAPI modes. This has made asio4all totally obsolete. The argument that new users need it is totally unfounded. Why? Just last week or so I installed Sonar on a W10 Laptop. Sonar worked right out of the box and checking in preferences it was using WASAPI shared mode. So why would a new user need a 3rd party lower quality audio driver? Sonar/ Cakewalk are tested with WASAPI 100%. And if there's no ASIO driver installed it knows what to do. It seems the only time people end up using asio4all is because some Chinese manufacture's of crap interfaces sometimes recommend it. They should have tried WASAPI first! If a Audio device doesn't support WASAPI it's a total POS. Toss it out. But that said it is true that Cakewalk/Sonar is possibly the only Daw that barfs on Generic ASIO drivers.Last Summer I had installed Cubase demo so of course that gets you the dreaded Steinberg Generic ASIO driver. At the time I was avoiding Sonar as I tested a bunch of other DAW's. A month probably went by using a few other Daw's without issue, They all used my Motu interface. Actually only Cubase asked if I wanted to use it's driver which I ignored and didn't register what this meant. I was totally unaware the Generic driver was installed until the day I opened Sonar and noticed it had taken over! Go figure.