RexRed

Perhaps, the biggest improvement of the VST3 plug-in is that it doesn't waste CPU resources and only works when it detects the presence of an audio signal, unlike VST2, which remains active at all times. For users, this means an opportunity to use a bigger number of plug-ins without overloading the system. Google https://blacktidemusic.com/software-and-plugins/vst2-vs-vst3/ Another article that does not mention bitrates or conversions of these interfaces. Melodyne definitely resamples the wave, this is the reason why it works on a 32 bit file in my project. What bit it resamples the wave to is still unknown. Considering ProChannel is on all the time this could account for CPU usage and load balancing may address ProChannel's use on only one core. And just because a plugin is VST3, this does not necessarily mean that infrequent use is implemented in that particular plugin. Then we have multi function plugins like Izotope Ozone 10 that seem to rely on many inputs. It would be logical to assume that no conversion is happening when a VST plugin is used. But ARA definitely converts/resamples the wave file. I recall the old ARA that left a ghost file of the original wave that was muted. And Melodyne must be resampling the wave to a lower bit depth if it cannot work on 32bit waves yet it does seem to work on them. Does it resample the wave to as low as 16 bit? Still unknown. 16 bit is not out of the realm of possibility.

https://en.wikipedia.org/wiki/Audio_Random_Access In this whole Wikipedia article and all the links at the bottom, there is no mention, not even a whisper, as to what wave conversion process is going on under the hood with ARA2. The DAWS are evidently passing the wave files off to Melodyne and Melodyne is resampling them. Am I sending off to the music stores 24-bit 96khz song files with 16-bit resampled vocals? And what about effects? Are some VST effects forcing my wave files down to lower bitrates? VST2 vs VST3? VST2 VST3 and ARA ARA2 are interfaces. The question becomes the quality of these interfaces. Another question, does Pro Channel also rely on VST?

I am guessing the ARA interface is 16 bit. This would explain a lot. I found the PDF file as well, thanks for the heads up on that. I asked if variable bitrates would not help with the bends of notes/blobs. My email from Celemony ************************ Hello Rex, thanks for the mail. Melodyne does not support 32 bit recording and variable bitrates. I doubt it is in any case halted to transients missing. The bitrate sets the resolution of the amplitude at a given time. I do not know of any recording software that would assign bit in that way. With best wishes, Ulf Kaiser ************************* Ulf has been a great help every time I have had to reach out to him. In the next email he sent me links to some tutorials for Melodyne. They were very helpful but it became clear that Melodyne is really only supposed to be applied to very specific parts of notes. Namely, the part where the main tune is, omitting the drift up and from the note/blob. Therein lies the rub, when you apply Melodyne to a section of the blob where the pitch "bends" this is where you get the wonky sound. It distorts the tone in a boxy and unflattering way. So it is either, get your bends right or get your bends right. Melodyne won't help much on those areas. Then I got this email message. ************************** Hello Rex, there's no need for 32 Bit recording or 96 kHz. A proper voice recording will do fine. The rest needs to be done using the tools in Melodyne: Separation, Note Assignment and Sibilance etc. Make sure to watch a few videos here, it might help: See: https://helpcenter.celemony.com/M5/doc/melodyneStudio5Training/en/M5tour_Videoseite?env=proTools Producing skills: Lead vocals https://helpcenter.celemony.com/M5/doc/melodyneStudio5Training/en/M5tour_Videoseite?env=proTools With best wishes, Ulf Kaiser ************************ While these video helped a lot it also highlighted the fact that many parts of your recording cannot be altered with Melodyne. This brings me back to the question. If ARA 2 is only 16 bit conversion this might explain this limitation. Ulf says Melodyne does not support 32 bit recording. What he does not say is whether if ARA 2 supports even 24 bit recording. As for needing 32 bit Support in Melodyne I can see how the tools in Melodyne can be utilized to a great extent to mitigate the tone corruption of blobs. But I wonder if having Melodyne applying itself to higher quality wave files would not also be beneficial. I am wondering if signing my lines in key is not simply the best option.

I am not sure this worked right. It took me to a help file about, "How to use real-time effects". I can see how Melodyne is sort of a real-time effect. I think this is using effects on the clips bin. How do I find pdf p1013?

Thanks for the info SCook. A link would be awesome. ? When I right click on the audio clip and select associated audio files it says, bit depth 32.

Okay, I found that, I know I have been told that before but I forgot it. Why is Melodyne working on this 32bit wave file? Is it converting it to 24 bit? I would think Melodyne would give me an error and say, "This is a 32bit file". Is Cakewalk playing both 24 and 32 bit files in the project?

I start a new project and record a segment and the project says 32 bit, then I close and reopen the project and up in the play transport under the now time it has changed to 24 bit. But my interface says it is running at 32-bit float The preferences say 32 bit. Then when I go to put Melodyne on the recorded 32-bit file it works. Melodyne is not supposed to work with 32bit files. What is happening here? Is Melodyne converting the file to 24bit? Are 24 and 32bit files both working in the same project?

Experience is certainly relevant. I just like the idea that if I need pitch correction on a certain syllable that it actually works and does not produce a wonky tone and not work. Having tools that work only part of the time is not the best way to accomplish things. When those tools actually degrade your work then it is worth trying to improve the quality of those tools.

I think they called that "more excitement" back in the day, "hype".

If that was the case they would have used only one take... Yes, critical thinking and rudimentary deduction is required. Do you even have any experience recording with a reel-to-reel? I recorded about 200 songs on 8 track reel-to-reel in my younger days. "Reductio ad Absurdum" aptly describes your criticism with pitch correction.

That response perfectly matches your knowledge of this subject. Excerpt Variable tape speed By the 1940s, studio engineers could produce primitive pitch correction by tweaking a reel-to-reel magnetic tape recorders varispeed. This process became more popular in recording studios during the 1950s and 1960s. By slowing down or speeding up a part of a recording and splicing with the tape containing the majority of the song, engineers could alter pitch. Another method of varispeed pitch correction was to slow a tape machine down, re-record a new part at a lower pitch, and then bring the recording back up to its original speed. One of the most famous examples of varispeed pitch correction is the recording of The Beatles 1967 single “Strawberry Fields Forever.” The song as we know it is not one single take, but two takes (7 and 26) spliced together by producer George Martin and his innovative engineer Geoff Emerick. The Beatles had recorded these two takes at slightly different tempos and pitches. By speeding up the slower of the two tracks, Emerick was able to match the tempo and pitch. https://www.izotope.com/en/learn/a-brief-history-of-pitch-correction-in-music.html It would be nice to have a producer or two and and few "innovative engineers" helping me make my music.

Your truth is rather deluded. The truth as you see it from your own personal but apparently limited perspective. The old recordings were pieced together like a jigsaw puzzle from spliced segments of endless takes taped together by engineers who were paid handsomely to make the singer sound perfect. If you like that then fine, go for it. Just don't expect us to swallow your bias as it if is "truth". ...and sincerity will get you a bad cup of coffee. The old performers would have sometimes ten engineers in the studio making their music while today we do it alone. There is something about music that is not quite so schizophrenicly produced.

What you say is true but it is also insulting. Everyone uses crutches. Did you build your guitar from scratch or did you buy one made from someone else? Did you make every sample in your song of did you buy some from others Did you build your micro chip in your computer yourself? You can't claim purist whenever it suits you while ignoring the elephant in the room. There are reasons for repairing a take with Melodyne rather than re-singing it. Because the take flows better and is more human that bits and bits of takes that were punched in. Did you even consider that? This is a discussion about gear and how to get a robust signal. You can apply Melodyne as a blanket and then uncover/undo the bits you need to as well. Your concerns are noted but if in the end a good song is created then your concerns are rather moot. Did you cut down the tree and collect the graphite for that pencil you wrote your lyrics with? Then you sir are the one with the crutch. ...and a bit of hypocrisy. ?

I have been contacted back from Celemony. Anyone interested in what they had to say?

No, you don't see... and that is the whole point. If you want hurl insults go find a troll site. Patsy Cline also scored some one one takes and your point is? She didn't score those takes on every song. And that was when they rewound the tape and said, "Try that line again Patsy". And she resung it and resung it until she got it right. Your problem with Melodyne is very uninformed. I can pick the bad tuning out of all of those old singers recordings you have riddled off. They were not all perfect but good enough to pass. And, while you were snoozing, even Madonna's old songs were autotuned and rereleased. This is not about philosophical justification for Melodyne it is about getting good and robust recording performance out of an audio interface. Please stick to the topic and keep your chip on your shoulder vulgarity to yourself.

I just emailed Celemony with a few questions. I have a question for people here, can Cakewalk record with 24 bit variable bitrates? The problem I have is at the beginning of notes. I don't know if this is a latency issue or not, but it seems the beginning of notes are (I don't know how else to describe it) paper thin, meaning, it gets recorded okay but if you try and edit the pitch, the tone disintegrates. It is as if not enough bits were used to listen to these crucial moments of the recording. This reminds me of video editing where a noisy part of a scene will occur, like an explosion or a foggy scene and it will require A LOT more bits to reproduce or copy that section of the scene. Well, this is the way music is as well. Vocal recordings do not respond well to a one size fits all strategy. Certain parts of the music need more bits. In practice, Cakewalk should throw more bits at the transients of a signal. I have a certain singing style where I throw my voice at the beginning of notes. It creates a weird pitch acceleration at the forefront of a syllable. This pitch part needs more resolution and something, I don't know if it is Melodyne or Cakewalk, but it is treating everything uniformly and not paying special mind that these parts of the sound require a much higher resolution so they are more manipulatable and refined compared to other parts of the sound that are not as complex a wave quality. I have been struggling with this issue for many years and gone through dozens of audio interfaces looking for one that would reproduce my sound more robustly. This is just another attempt to figure out what I might be able to do to get more bits into Melodyne. Is this part of the ARA 2 standard to treat all sound uniformly when converting it?

I just ordered this interface. I need to find out how well this works. I will let you know what I find out.

Every professional singer in the industry uses Melodyne or they sing the same line over and over 50 times like they did in the 50s and 60s till they get it right. The wonky syllables are the nature of how pitch correction works, it only works on certain syllables and does not work on others.

I wish it was a singing issue.

I am thinking of purchasing this audio interface. Zoom UAC-232 USB 2.0 Audio Interface https://www.sweetwater.com/store/detail/UAC232--zoom-uac-232-usb-3.0-audio-interface It says it is the first 32-bit audio interface even though a lot of them lie about it being able to actually "record" in 32 bit. I hate every audio interface I have ever owned. I love my RME Fireface UCX 2 but I hate the audio quality even though I like it much better than all of the brands I have owned previously. It gives me the ins and outs I need in a modular way. Everyone says that the only difference with 32-bit recording is headroom. It is hard to believe that recording something with tons more bits would not produce a better-quality recording as well. Or let's say, a "more stable" recording. Question 1. Will Melodyne work with 32-bit files? 2. Will my effects work with them? 3. Will my VST instruments work with this interface? I can record my voice and it sounds perfectly realistic but the moment you move some syllables in Melodyne the tone gets all wonky and formant does not come close to fixing it. With more bits you would think you would get a more stable representation of the sound and be able to modulate it without the syllables disintegrating. Any thoughts? I have gone up to 96 khz and this did not fix the wonky syllables in Melodyne problem any better than 44 khz. Going up in khz there was zero improvement in syllable stability. My thought is throwing a billion more bits at the problem would fix it. I would like to hear how this sounds and so I get bigger files and not being able to distort my input that is definitely a plus as well. The advertisement for this states that not only do you get better headroom but also better "quality". Quality is the thing... And about Cakewalk, would these 32 files bog down Cakewalk playback? I have 18 Intel extreme cores, will a 32-bit song limit me to like 4 tracks with only a few effects or something? Would things like vocal rider still work? Thanks in advance for any input on this. I would like to try this, maybe if it does not work well I can return it.

I was once told by a sound engineer that 70% of all room acoustics come from the border between the ceiling and the walls. Put your treatments there. A couple years later this same sound engineer introduced me to Cakewalk for MSDOS. I know you are all going to cringe, but I use earbuds to do 70% of my mixing, not headphones. I like the direct connection to my ears they provide. 90% of my listeners use earbuds. Headphones to me sound sort of dull and I have had some good ones. If I use headphones, I like Philips brand, and they will run you anywhere from 20 to 40 bucks. Sonarworks has my headphones tested for the frequency curve Philips SHP9500. But, I still prefer earbuds. You MUST use headphones/earbuds and monitors to create a mix. I consider headphones and earbuds to be pretty much the same thing even though I have my preference . There are things you can't hear or judge in headphones and things you can't hear from monitors. These are the things you listen for when using them. Earbuds will reveal if my bass is just loud enough to rattle the bones in my ears where studio monitors will mask that. In earbuds, my vocal I can crank way above the music and the song will still sound fine (they lie), but when I use my monitors, I can precisely set the ratio of vocal to accompaniment. I will put it this way, it is impossible to set the ratio of the voice to music in earbuds. Where, monitors tell you immediately if the voice is too loud. I use M-Audio Studiophile BX5 monitors. I have big PA speakers too, but I don't use them anymore except for live gigs. Earbuds are like a microscope, you can zoom way in with perception and fidelity and detect even the slightest noise where monitors you zoom out and you can perceive the scope of the music. Many times I have mixed a song in earbuds and convinced myself that the vocal was nestled perfectly into the music only to be horrified to find out the vocal was way too loud, or out front. This can happen after a song has been published, and it can become a source of embarrassment. So you use each for different perspectives on your mix. You eventually learn what to listen for with each. It is an uncomfortable feeling having to switch listening sources when you have become intently attuned to one for hours while mixing a song. Never release a song without listing to your song in both. Reverb sounds different in each, so you have to add up the pros and cons of this phenomenon. I use earbuds mostly because I can hear many minute and subtle things that are masked by monitors. Once I set all of the minute things right, usually by that time, I think the song is done. Only to find the monitors say, not so fast... I use cheap Panasonic earbuds, I think I paid 6 buck for them on Amazon (buy several pairs). https://www.amazon.com/gp/product/B07J4WHNFC/ref=ppx_yo_dt_b_search_asin_title notice there are over 100,000 ratings. That means millions have been sold. I believe this product may be the largest selling product in the history of Amazon. I use them when tracking as well. I can wear them all day (and into the night) with zero fatigue. ? Hope this is some help. And, you need to cut some bass out of everything (prochannel works great for that). Oh, and shut the doors, an echoey hallway can ruin a recording.

All awesome suggestions FreeEarCandy, I just spent the last 5 hours working on this song. After finishing my changes I came here and noticed your post. I hope the changes I made are in concert with your ideas. I did close to a thousand changes or more. I put the guitars in phase... It was so all over the place that I could not get the song to mix. I think it is better now. I did try and make subtle changes and shifts so the only thing that plays throughout are the rhythm guitar parts which I try and mask a bit here and there. The harmonies are subdued, I added new pads and took out the brash brass and replaced it with some wet warm strings. And an Omnisphere pad and other pads. There are several pads that take turns. Added the Oos at the end as well. I added a guitar solo here and there. Re-sung a line. Added a ton of effects to everything. Did a lot of lead vocal editing. It is not as acoustic as I wanted it, it is more new age with a synthy feel. I call this genre, "folk rock". Quite a stone's throw away from the urban hardcore punk rock that I usually do. lol ?

This is an older song I made, I spent a couple days revamping it to get better sound. Any suggestions? Anything that sticks out? Comments and critiques are welcome. Thanks for any input! Best! RexRed

During the verse there are "dubs" where I sing certain lines in unison. They are a bit strong. I will look into toning them down a bit.