Gswitz

If you can tell visually, you have to play through the track. Cakewalk is not great for this purpose.

+1

https://www.researchgate.net/profile/Amandine_Pras/publication/257068631_Sampling_Rate_Discrimination_441_kHz_vs_882_kHz/links/00b7d52459de5acbf6000000.pdf I was looking for info for azslow on stereos reproducing content above 20k and how good of a job they do. Found this interesting.

I guess, Will. I'd have to see the implementation and give it a try to see how i like it. It's hard for me to imagine a computer generated clip gain envelope that i would feel comfortable using without veto of each decision point. Not even seeing the envelope... Idk. Hard to imagine using it.

2 knobs help get levels right. Expensive pres sometimes come with gain and trim. They do the same thing but at differing degrees of impact. One gets close. One gets it just right. No shame. Use them. Forget your color worries.

If you have the envelope present, you can swipe a range, hover high-side, then drag up or down. This will auto create 4 nodes around your selection. Then adjust slopes if you want.

Clip gain is an envelope. You are in total control of the regions to raise and lower. All the usual tricks work. You can easily raise soft words or cut breaths. It gives you nice automation control of gain before your fx on the track. I'm not talking about track gain which is a static setting on the track. Take a look. It is more handy than i knew when starting out. Glad I've discovered it now. Arguable drawback is you don't get to see the waveform change as you adjust, but by doing a bounce to clips followed by ctrl+z you can see what it does to the clip briefly prior to hitting the fx chain.

I think that's it in a nutshell, @micv.

Hey Azslow! With lasting respect for you, I took a minute with my RME UCX to do the following to show that RME exports the higher frequencies to the speakers... 1. Dragged a loop into a new project at 96k (shl_guit_80_ringshft_D_Rex(9)) -- noted audio ranging up near 20k. 2. bounced to track to get an audio file I could use the loop construction view on. 3. Used Craig's Les Paul trick from p270 from the Big Book of Sonar Tips to speed it up 7 half steps. 4. sent the stereo pair out of the interface and back in physically. 5. Recorded the raised recording and noted content up to around 40k. 6. Next, I slowed down the new recording by 7 half steps and did a null test with the original. It was pretty good to around -33dB. So this shows that through this process, most of the audio was retained. This defends that the RME UCX will send the higher frequency content to your speakers. If you were questioning whether my speakers do a good job up there... haha... not worth testing. 🙂 I have cheapie charlies... but I'm sure you can find speakers that attempt it anyway. As an interesting aside, while the first and final spectral analysis bits are almost identical, at one moment you see a tiny jump at 40 (otherwise quiet between 20k and 40k like the original). So, not perfect null test as I said. Something funny in there. No other gear or interface in the middle. Out of the RME UCX on channels 1 and 2, in on channels 5 and 6. No pres on those channels. Because that final 40 was after slowing it back down, I suspect something to do with the loop construction view slowing algorithm. Who knows what. 🙂

You can automate clip gain and then bounce to clips to materialize. If you like what you see you can ctrl z to return to the clip gain envelope in case you want to change after listening. I leverage clip gain A LOT. It is useful for gain staging into fx. And when mixes get hot on a bus, you can back down the bus gain.

I think the gentler filter is the best argument for 48 imho. For those who don't know, if you play back digital audio with content above the nyquist frequency you get artifacts. That is partly why 44.1 was originally chosen. Half of 44.1 is 22.05. To avoid aliasing noise on reproduction, a sharp low-pass filter removes content above 20k to nothing by 22.05k, just above the scientifically asserted upper edge of human hearing. So, when you playback the file, there is nothing at all above the nyquist frequency. This sharp eq has its own impacts. Doing the same between 20k and 24k is a more shallow filter and less discernable. The 24k ceiling is for the 48k rate. At double and quad rates it hardly matters where the filter starts as long as there is one. If you don't use the filter, the aliasing is clearly in the audible range double rate or no. You do not apply this filter. Your interface and daw do it. The quality of this filter and how it is done distinguish the elite interfaces from the rest. To see if your interface works properly at higher rates, use a spectral analyzer that shows the whole available spectrum and record your mic at 96. See weird stuff up near 40? I have seen several interfaces including Audient that tried a poor filter at 20k when doing 96k recordings. There was tons of noise then bouncing to the max signal level of any frequency near the nyquist. By this i mean, the lines between 20 and 40 were mostly quiet but near 40 there would be as much noise as the noisiest audible frequency. Sent it back twice and it now works properly. Records dog whistle pieces beautifully. There are other interfaces identically goofed so it must be a common engineering mistake. For fun, you can use an eq to remove all content below 20k. Bounce. Then lengthen the recording without snap points for a perfect stretch. This will bring that content to the audible range for your enjoyment. You can also do the eq after the stretch instead of before. The benefit of before is that you know where the 20k point is. 😁

I mostly agree, unless you are letting each instrument record unlimited takes and planning to keep them all in the project until comping. I hit my head on that ceiling still and my pc isn't shabby.

I think 48 is more than placebo myself because the anti aliasing filter can start higher above 20k and be more gradual. But what evs. Bit, i know you were talking about double rates. I do often track at double rates. Makes sense to me and gives me piece of mind. I'm ok with the idea that maybe we can detect frequencies above 20k. So, to preserve those, you would have to track at double rates, mix stems and master at double rates and distribute at double rates. The bands i record never ask me the rate i track at. But if they thought there was $5 in it, they'd ask for a double rate export. He he. (By this i mean, pop open that pricy download and look for content over 20. See any?) If done correctly, a 96k file can have content up above 40k. See dog whistle. Quad rates required for a proper cat whistle 😜.

This is true of the bounce happened at a higher sample rate, but i don't think that is what the op is getting. Project at 48... Saturation plug Export audio to 96... ... This will result in the saturation being processed at 48 then the result converted to 96 with badness printed. Yes? .... On the other hand, using cakewalk up-sampling to 96 for the plug and exporting audio to 48 avoids the badness of the plug. ... No harm is done exporting to 96, but unless i misunderstand, i don't think there it's much potential benefit. Perhaps if he has an interface from the 2000s the higher rate might reduce issues related to the interface.

Because of video, for some years i have been archiving 48k 24 bit files. My phone plays them fine, which i use to listen to mixes on my commute to my day job. I haven't burned a cd for myself in 2 years. I think that CbB bounces to 32 bit by default internally. I allow this. Internal bounces have never driven my io out of range. I can surpass io limits when practicing a tune a couple of hours a day. The takes add up and require deleting from the project. I don't know why i record practice. I guess it is so as i develop as a player for the lead, i can then switch to develop my comping and have a better lead to work with and vice versa. I don't think you can do internal bounces to a different sample rate than the project. You would have to be doing an export to change the sample rate. I always do an internal bounce prior to export. That way i can keep versions of the mixes as i work my way through them. I end up with a stack of mixes to compare and choose from. I save scenes for each mix. When the artists swing by, they can get a 10k foot view on each version of the song to quickly assess the mixing choices and veto things they don't like. You can use exclusive solo to flip from version to version helping them hear differences in equally loud versions. If you export your mix without an internal bounce, comparing different bounces would be hard.

Recording at higher sample rates gets you more headroom with equivalent fidelity. Most vsts for audio will sound the same at 44.1 as at 96. Using all available bits (loudest moment in the song hits 0), 44.1 can theoretically reproduce everything a human can hear up to around 108 decibels at 16 bit. At 24 bit 44.1, there is enough information to reproduce everything a human could hear if one could listen at 160 dB. Thunder is 120 dB. (This doesn't guarantee your stereo does a good job of reproducing.) Now soft synths are different. Guitar amp simulators may be different. Those may audibly benefit from a higher sample rate depending on how they are written. To this end, you can choose to have cakewalk run any vst at a double rate without altering the project. Bouncing a 44.1 Recording to 96 will likely have no detectable benefits imho.

If you have a battery operated cable, check that first.

Ah, gotcha. That's different. I haven't had that particular problem. I'm sorry. I can sympathize that it's frustrating when you can't get a midi signal and it's Cakewalk's fault. You do a lot of unrelated troubleshooting before you id Cakewalk as the offender.

@Rico Belled Right! I think the problem is being driven by a failure to understand that your midi input is being bound to a controller by default because you have a controller specified and the original midi input for that controller is not available. In this case, Cakewalk binds a different midi input to that controller and doesn't tell you. I don't think it even impacts the project until you close the Cakewalk and re-open it, making it hard to know the cause. From Cakewalk's perspective, it's working as expected. I challenge whether it SHOULD work that way. For me, if the midi input I selected for controller is not available, I don't want Cakewalk to use on of my other midi inputs for it. They aren't controllers. 🙂

For me, this is almost always caused by cakewalk having the midi mapped to a controller in preferences. If you have a midi controller... Then unplug the controller... Cakewalk will assign the next midi input to that controller on cakewalk restart. This prevents you from using that input for a midi track but not from selecting that input on the midi track. Additionally, it separates you in time from the triggering change, making it harder to associate the action with the outcome. Personally, i think this is a use-case-bug. Perhaps not an actual programming mistake, but a use-case that stymies users from success using cakewalk. When a user attempts to record midi routed to a controller, the user should get a warning like when a bus is routed to an output that doesn't exist. I find it necessary to delete controllers in preferences when they are not plugged in or leave them plugged in. It is a bit of a pain. Every time i plug in my controller, there is a bit of configuration i must repeat.

Eq is a matter of taste. It takes time to learn how mixes sound on different systems without checking them all. This is an art. You don't distinguish yourself by copying others but you can hone your skills that way. Play with presets and spend some time playing with the fx. Try sharing a mix you think is garish and see if your friends agree or not.

New guitar strings.

Complaint submitted. 😜

Keep at it. You disabled usb select suspend in advanced power settings? Removed unnecessary devices? Click start button, type background, click disable all background services. In latency monitor, what is still listed in driver tab at the top sorted descending.

Usb is showing up in your report. Are you sure that you have disabled select suspend? How about power save on usb hubs in device manager? Unplug all but your interface, mouse and keyboard. Is your keyboard wireless? Windows key+x... Select device manager. What is this coming from?