RexRed

Yes, I have noticed the same thing, am having trouble getting my drums to fit in some of my slow songs. I thought of leaving them out but I left them in to add some interest to the scant arrangement. I am thinking a brushes might work better on the snare than sticks The HH tapping is part of the one of the guitar waves I bought on splice.com and I cannot remove that. They decided to add that tapping to a guitar chord part, it is not something I can remover or attenuate.

I am going to try recording a project at 32bit with 48khz I am assuming it will record the tracks at 32bits at 48khz. This logically means in my thinking that the 48hz will be subdivided up across 32bits. I am assuming with more bits there will be much more resolution. I may need teo learn how to downsample the effects until the last minute before the final render. I am hopping this will solve my problem with fuzzy transients in my Melodyne blobs. The performance of the formants tool in melodyne is really poor and I am trying to fix this problem and I am hoping the throwing more bits at the problem will do it. I will let you know how this all turns out. I kept it at 48khz because I think that audible range is enough. I am excited to try out the downsampling effects capabilities in Cakewalk because I have a suspicion that at 32bit it will bog down my CPU even with 24 hyprethreads. Projects capable of using varying sample rates would be a very nice feature and perhaps give Cakewalk yet again one more edge (along with many others) over other DAWS .

Thank SCook for bringing this awesome feature to my attention, I briefly heard about that once before and had not really thought about the implications. It would be nice to be able to lower the sample rates and bit depths of certain tracks so I set the project to a very high rate and then decide as I go along which elements require that kind of precision. Maybe if these elements could have backup tracks so they resort back to the original rates when rendering. I am not so much concerned about project file size as I am about real-time playback CPU load as I like to mix and master simultaneously.

New song Cakewalk peeps! Hope you like it! Best to you!

Years ago I would have to program the drum machine and record that track first on my reel to reel before I would lay down any other tracks. So every drum roll and cymbal crash had to be preplanned out before I would lay down any other instrument. Gone are those days. Now I can just place a drum in any place in the song as I go along. What a change that was when technology developed enough to do that. I seem to be at that same crossroad but now it is with the music and vocals. I want to make the music lower resolution so the effects overhead is at a reasonable level. And I want to make the vocal track at the very highest resolution possible. This would require that I make the music beforehand and then import it into a project with a higher bit and frequency. Then lay down the vocals. I am wondering if there is a DAW out there that allows for different bitrates and frequencies to run simultaneously. Can Windows even do this and can ASIO and any Audio Interface do this? How about using something like VoiceMeeter Potato? I am thinking two instances of Cakewalk in sync to two PC's in sync... Sound cards slaving to the same clock or SMPTE? Any ideas?

Thank you Treesha for doing your testing of this problem. I have tried raising the khz but not raising the bitrate. I wonder if 32tbit recording might help this. I wish I knew more about this problem and how to remedy it. I think a variable bit recording might help this but how that would affect real-time monitoring while recording might be an issue. I am not even sure if I can record in Cakewalk in 32bit my Steinberg UR22c is supposed to be 32 bit but I am not sure how that is applied and what about Cakewalk 64bit precision how does that fit in. And can Melodyne handle 32 bit recording? I am a bit overwhelmed by this and my song files would be a lot bigger I guess another 12tb hard drive might be in order. I will use the 12tb to store projects and the M.2 to record and edit projects (cloud storage is gonna love me). And with 32 bit recording my effects/CPU might get bogged down, maybe I'll upgrade my 12 core to 18 core. Here is where having my Nvidia 3090 assisting in the record/playback/rendering process would help. I wish I could set the bitrate on a per track basis. Some tracks are background and some are up front. Just like in 3D art, background things can be low res and still be okay. Each track needs to be its own project layer container. Vocals up front and solo instruments would get the royal treatment.

Yes, that is exactly my experience also Marled... When notes jump a lot there is not enough bits thrown at them. It needs 2 pass variable bitrate, just like film clip rendering when there is an explosion or a lot of action in a scene, more bits get thrown at those sections...

Hello Treesha, again thanks for the comment, I rarely if ever use Melodyne on a single vocal clip, I usually apply it to all of them at once on a track. The only problem you can have doing this is Melodyne does not like it if the clips overlap one another. Go through them beforehand and make sure they do not overlap, they can be in separate take lanes when you apply Melodyne globally but they should not overlap or be doubled. You can also comp Melodyne, and delete a section within a Melodyne instance. You can place another clip to fill the gap and apply Melodyne to it (make sure it does not overlap). Sometimes Melodyne will still see the old blobs from the deleted comp section. Just save Cakewalk close it and reopen it and the old blobs will be gone.

Thank you for your response Treesha, drag and select several clips at once in a track, then right click on them and select region effect > Melodyne. Notice how Melodyne puts them all into one Melodyne instance. Then right click on that instance and select region effect > remove Melodyne. You will notice you are not left with several clips but one solid clip. In order for Melodyne to make them into one solid clip, my reasoning is, there has to have been some sort of bounce. When I make a vocal, I record ten to twelve takes or more of the entire vocal in a track, then I comp each part I want consecutively in the track, then I delete muted takes and I prefer to move all of the takes manually into one take lane so there is no resampling . Then I select all of the takes in that one lane. Then I right click and add Melodyne to the whole region of clips, I leave Melodyne and do not freeze the track so I can always go back into the song project and tweak the vocal in any place I notice a problem in timing, pitch etc. I save the project prior to comping with a new name, then I save it again with a new name after I delete the muted takes, save it again with a new name before adding Melodyne to all the clips and I save it again with a new name after adding Melodyne to all of the clips. I never freeze my Melodyne tracks. I do freeze my midi tracks almost always but rarely if ever bounce my Melodyne tracks. This melding of clips together in Melodyne I would refer to as resampling or bouncing. The fact that Melodyne can undo all of its edits and provide you with the original wave form does not imply to me, in my mind, that the wave was not resampled initially.

MSMcloed, Treesha and Mettelus thanks for you input into this discussion. This is in response to all of your helpful advice, I am using the latest version of Melodyne Studio. It is good to know that ARA is not resampling my waves but what is odd is if I remove Melodyne from some wave files it does not restore my exact takes... It leaves behind what looks like what i would get if I bounced multiple clips to track. So if ARA is not doing this then Melodyne is resampling my waves. About the algorithms in Melodyne there is only one algorithm choice for what I am using Melodyne for (melodic). It only stands to reason that something is converting/resampling the Cakewalk takes/clips into another file. If this was not the case when you removed Melodyne it would restore the takes/clips exactly the same way they were originally. It is this resampling process that I would like to know more about. It seems to be only one flavor and Melodyne only works with that kind of wave file. If this was not the case then higher project resolution would yield better pitch and formant editing fidelity. Please correct me if I am wrong. The issue I am having is I bend my notes up I drift up to my notes during the transient phase of singing. This is the area where I need better resolution in the resampling process. Melodyne does convert these okay but there is no deep resolution there to allow for pitch and formant editing. This does not always happen but sometimes even the slightest pitch or formant editing of the transient makes the quality of the transient sound wonky or fuzzy. This has been happening for years all the way back to VVocal over the course of three or four operating systems and many different computers. I would like more flexibility on what kind of resolution, filter or frequency response is being used within Melodyne to convert these waves. It is apparently not editing them in place or removing Melodyne would restore them back to their original state. It seem quite probable that Melodyne is converting them to some sort of generic format that it can understand. This conversion process I would like to have some flexibility with. I am sorry if I am not understanding how this works...

I keep thinking that I am not very pleased with the way ARA 2 converts my tracks into Melodyne. Is ARA downsampling the files? Is there a way to tell ARA 2 to do a better job converting the waves into Melodyne? Sometimes you can't move the pitch or formant on a syllable (especially a transient) even a tiny bit and the wave gets all "fuzzy sounding". You are stuck with the way it is. It seems to be the way ARA converts waves. It doesn't matter if the wave is 48 khz or 96khz and 16 bit or 24 bit. I am recording in 48khz 24bit, you would think that raising the khz or bits would allow you finer adjustment on the pitch and formants without the sound going all "wonky".

Great info OPunWide! Welcome to the forum! That makes me think about something Hatstand. How do you find out what is on your bus? I currently have all of my USB devices plugged into a PCIE card so my sound card is not reduced in speed. But, wouldn't a mainboard's USB 3.1 or a USB 2.0 slot also be on different internal busses? It does not seem likely that by using a USB 2.0 device it would slow down the USB 3.0 or 3.1 slot. It seems each generation would have its own dedicated bus. How do you find out which USB slots are on which bus? And how does this all figure into lanes? I have 44 lanes but I also have two PCIE 16x graphics cards. And how much of a limitation is the USB power? Is it enough were they have to limit the amount of chips they can put into the USB 2.0 AI?

Thank you for your reply MsMcleod, wow, so you are saying that USB 3.0 is really not so important. I figure (correct me if I am wrong) the effects are all done by the CPU before they are turned into a wave that is played by the AI. Also, I use livecast to broadcast video and audio over the internet in 4k and I hook to another streaming computer also. It gets complicated for sure. But it still seems if the AI was doing a couple stereo streams both ways in low latency it will still not reach the bandwidth limit of USB 2.0 I do not like that I have to crank the microphone gain so much on the UR22C, that bothers me. The meters on the Motu would give me a clear representation of the signal, this is also attractive. I also have heard that the UR22C is very noisy compared to other AI's but still not perceptibly. If I can eliminate the USB 3.0 factor, that gives me the freedom to go with the Motu. I am still going to keep the UR22C but I want to try the Motu and see if the preamps have a better impedance for my microphone that seems to be gain hungry. My headphones are also gain hungry... I don't like that I have to crank the headphone and microphone knobs up most of the way.

I have noticed very few audio interfaces have been switching to USB 3.0. There has been ample time for them to do it yet there are very few out there. I have to admit I am unknowledgeable about what is happening behind the scenes with my audio interface. I just plug it in and it works. There is so much else to concern myself with. But, is getting a 3.0 interface really worth it? I do not record more than one or two tracks at a time at 24bit 48khz. But I use a lot of effects and often have a lot of tracks playing at once. Where exactly would the bottleneck occur? I have been learning that audio interfaces differ quite a bit from one another. Everything from preamp microphone/headphone impedance, frequency response, and noise floor etc. Is USB 3.0 really necessary? I have a new Steinberg UR22C USB 3.0 interface and I noticed I have to crank the microphone knob up quite a bit. The UR22C is USB 3.0 but I am considering also getting a Motu M4x4 but that is USB 2.0. I like that the Motu has an actual meter on the front and it has a lower noise level and I assume it will also have preamps that accommodate my microphone/headphones better. Can someone give me the ins and outs on what the CPU does versus the interface and why and when 3.0 speeds would be used or necessary? I use a ton of effects and I also master at the same time that I mix. Thanks in advance for any help on this inquiry.

Thanks Marc for the compliments on my songs and voice. Glad you like them! Super you like the synth arpeggio riding the song out too! ??✨

Thanks for the comment Noynekker, Yes, it is a bit brash in places. I am thinking of adding harmonies and cranking down the vocal volume in a few places to make it fit into the music a bit better. It sort of passes the way it is but could use some very slight tweaks.

Wow! Thank you David for testing this out on your speakers, very nice to hear this tune held up!

Hey Wookie, I use my big 15 inch Kustom speakers to assess and bring the vocal into the mix and yesterday the transformer/amp circuit board in one of them fried. My studio room was filled with noxious toxic smoke. I just had them on and I was not even driving them with a signal. I am looking into replacing the parts so I had to guess the vocal to mix ratio through my headphones which does not work in my experience. I will revisit the song at some point. Thanks Jack for the compliment, glad you like it! ?? I thought the vocal was a bit hot and up front but it is in the ballpark I think.

Here is my new song, "Eternity". I hope you like it Cakewalk peeps.

Such a nice compliment Paul, I do take pride in not being contrived. I am glad you like this simple and easy style. ?

Wow Andy, The Eagles, I am very flattered! I grew up on them, I aspire be compared to such greats! THANK YOU.... ???

He Jack! The reverb is a REmatrix convolution reverb on a parallel bus using the JPVerbs "rich ballad" patch. I used the ProChannel compressor and EQ and Neutron 3 in the effects bin along with Waves Vocal Rider. I side chained the lead vocal so it would cut an impression out of the music tracks and sit better into the mix. And, Thanks Lynn for checking the mix on your systems! It is nice to know this song is translating well across other speakers! I hope this song does well, I am switching to a more pop style for the time being. Maybe a more universal appeal never hurt.

I like this! Especially when the beat comes in, robotic and mechanical in a good way. Excellent groove Wookiee!

Thanks Kurt for the nice compliment, (Dimension pro cello) I may add more pads to this and some subtle touches to give the music a bit more build and interesting flavors.

Oh, sorry for that Jack, I will fix it thanks for the heads up! Forgot to select the project before export. Yikes! Okay, the audio is fixed.