Peaky notes, EQ and compressor limitations, uncontrollable

Thanks - yes that was the idea but after doing a test run on my MP3 player I was not happy with the result, when I had to up the volume due to noise around me where I was listening— Such dynamics, I guess, only sound good when you have a nice quiet space and decent system to listen on. — SO I have redone my song Ultra Insanity, adding the usual High Pass and SC4 compression… It’s trial and error, as usual… And I’ll probably fix all my other songs ASAP and possibly post the details ASAP. Update: song removed

UPDATE - Still not liking my edit I think too compressed now! Stumped again - “back to the drawing board”

I don’t know if it will help or not but I listed some of the plug-ins I use in my post “multitrack recording I made using audacity” on the same forum. Maybe they will work better for you.
Also, I usually don’t use compression much, instead I use Nyquist limiter on the final 2 channel mix to remove the short peaks and then use Normalize to get the track louder.
Maybe?
Bob

Big Thanks Bob, I will check that out ASAP…

comment removed

Hello, forum - Is there a tool to remove vocal smacking sounds?

I record acoustically/unplugged and the mic picks up every little unintended smack from my mouth as I try to deliver vocals as intimately as possible, with much quietness in my style it is a problem I have been fixing with the Repair effect mostly but this is laborious and not always effective.

Here is a link to my edited/finished songs where you will still notice some ‘smacking’ here and there that I was unable to remove or may have missed: update: song removed/problem solved

Also, for recording flawless vocals, does anyone know what one may or may not do before performing to lessen this problem, like a certain beverage or food to use or not use? I don’t use dairy.

These are not ‘pops’ so I assume a pop filter will not be the thing to fix this.

Also, any kind of gate I know of will not work due to these smacks being so interwoven in the vocals where volumes of course vary.

https://forum.audacityteam.org/t/updated-de-clicker-and-new-de-esser-for-speech/34283/1

Koz

Thanks, great tool; Is there a manual for this anywhere?

This works great — and on my acoustic vocals/guitar tracks as well. Thanks.

It got rid of all /with only a few I had to do manually, saved a lot of work - much appreciated!! I used default settings and did not attempt to tweak this incedibly complex plugin — which frankly I don’t understand - most of / Regardless thanks for the recommend - and thanks to the creator of this great tool.

UPDATE: For my music editing I learned this is best used at the specific problem, on specific areas (where it can be previewed for accuracy first) and not on the whole song (or if so incrementally) because it will occasionally distort the sound. Regardless it is a good tool for music editing as well.

Peaky Notes Update Feb16 2017

Editing per one stereo track recordings – /acoustic guitar/vocals on one mic --. “Less is more” it seems for these. Even compression seems to ruin these delicate tones. On my new edit here, this one (As The World Fades Away):

http://www.soundclick.com/bands/page_songInfo.cfm?bandID=1397866&songID=13531763

I did one High Pass Filter (at 70 hertz/6 dBs) and then with EQ took nine decibels off the mid area between 160-250 hertz. Then just cleaned clicks and a few esses and repairs and that’s it. Here is the finished song’s plot spectrum/first 3-4 minutes read-out.
AS THE WORLD HP70_EQs160-250_-9.PNG
Is there a normal look or a right look to such a spectrum? Any spectrum? I use the spectrum to edit, sloping the low end down from the mid peak (the song’s peak), though this one was a bit fiesty even after High Pass Filter at 70 Hz/6 dBs though I decided to leave it (and not hit any heavier with the HPF, or with EQ) hoping for a little more thickness to the low end and not wanting to hack it down by any other means either, like Notch Filter. Then I reduced the mids highest section down by as much as it showed to be jutting out from the rest of the mid section, in this case 160-250 hertz down nine decibels.

NOTE: When I record, with my Tascam DR-05 I have the built-in limiter set to on which keeps any high volume input from clipping. I also record as 44 kHz/24 bit WAV.

Does anyone think recording in 24 bit is not preferable to recording in 16 bit, when the final export will be 16 bit?

Or which of these options is best for my type recordings/edits:

  1. record 24 bit, edit 32 bit float, export 16 bit
  2. record 24 bit, edit 24 bit, export 16 bit
  3. record 16 bit, edit 32 bit float, export 16 bit
  4. record 16 bit, edit 16 bit, export 16 bit


    Any recommendations appreciated.

UPDATE: I’ve switched sites around and at this time have a new mostly proprietary method of things (previous voided). If you like my new sound and want to know more about my editing techniques feel free to ask,

Ronald Newman
http://www.SoundCloud.com/BlackDogSongs

That’s a contradiction.
A high pass filter at 70 Hz takes 3 dB off at 70 Hz.

Thanks Steve… That paragraph corrected:

“I did one High Pass Filter (at 70 hertz/6 dBs) and then with EQ took nine decibels off the mid area between 160-250 hertz. Then just cleaned clicks and a few esses and repairs and that’s it. Here is the finished song’s plot spectrum/first 3-4 minutes read-out.”

I might have been high when I wrote that originally - on chocolate and air pollution.

[re] unanswered questions:

#1. is there a normal plot spectrum look? or just guidelines as to what the graph shows or should show/…and/or are there just recommendations somewhere? what it’s telling us/and where and when one might want to change something… based on it?

#2. (what is the best combination of these possible recording/editing choices when exporting as 16 bit:) “recording:16 or 24 bit/editing: 16, 24, or 32[floating] bit” — [again] when exporting as16 bit? what’s best? …

#3 At a certain point could there be editing/sound engineering questions that would be considered proprietary information (or secret to a select few)?

UPDATE: I’ve switched sites around and at this time have a new mostly proprietary method of things (previous voided). If you like my new sound and want to know more about my editing techniques feel free to ask,

Ronald Newman
http://www.SoundCloud.com/BlackDogSongs

I guess so…

Anyway I’ve got a few ideas myself but 'am starting to feel a ah “proprietary bug” … myself? (I guess/maybe?) … or just a, “casting pearls before swine…” kind of feeling. Did I just sum up the whole internet in it’s entirety? (or am I “…the kettle calling the pot black…”?). Or wait, maybe this is all just the newer vaccinations hastening some kind of zombie-like apocalypse we’re all just in the early throes of? … “Zombies” … how fun … (not!). I think I need to move (from Scottsdale Arizona, “The Chemtrail Capital of the World”). Surely the whole world hasn’t turned to zombieville… Okay will someone now please delete this comment it has nothing to do with… ah where am I?

UPDATE: I’ve switched sites around and at this time have a new mostly proprietary method of things (previous voided). If you like my new sound and want to know more about my editing techniques feel free to ask,

Ronald Newman
http://www.SoundCloud.com/BlackDogSongs

I don’t understand how to equate the concept of “skills” with the concept of “proprietary”. Most equipment and software include documentation (a user manual of some sort). Audacity has very comprehensive documentation (http://manual.audacityteam.org/ and Missing features - Audacity Support)

It depends on the type of material. To get an idea for a specific type of audio (say, “acoustic blues”) try looking at good examples of recordings in that genre.
As a very general and loose approximation, look at the slope of “pink noise”.
Generally, there shouldn’t be big spikes or deep valleys in the spectrum when averaged over a long period.
Very low (sub-sonic) frequencies are usually rolled off to zero at 0 Hz, though some genres may have substantial energy at very low audio frequencies.
The upper frequency limit of very old music recordings may be limited by the equipment used for the recording.
The upper frequency of modern recordings may be limited by the audio format (low bit rate MP3s usually cut off frequencies well below 20 kHz).

Best to edit in “32-bit float” as this provides best quality processing and avoids risk of permanent damage if you go over 0 dB during the course of editing (the exported 16-bit file cannot exceed 0 dB).

Zombies :mrgreen:

thanks STeve … re-calibrating …

QUOTE-from Steve:

Very low (sub-sonic) frequencies are usually rolled off to zero at 0 Hz, though some genres may have substantial energy at very low audio frequencies.

I’ve never seen this — “zero at zero” this is interesting/this may be why I seem to always be battling a low-end tone in my songs/song edits. I’ll have to look at/analyze some more popular acoustic blues songs to see this…(look for this) What I’ve noticed (with pop acoustic blues) is the lows are up but just sloping down from the mids high point of the song per frequency analysis, and of course the older blues much more cutting of the lows.

-Still frustrated!

I recorded my last song in 16 bit (“WTF”/explict, on my site, link below) just to see if recording in 24 bit has been part of my problem-- At this point I am unsure but it turned out much less bassy/muddy then all my 24 bit edits, not sure if a fluke at this point though… The only disadvantage with 16 bit compared to 24 seems to be the esses are worse and clicks harder to fix… (so far/and that’s a big problem!) But again unsure at this point. Could recording in 16 bit be better than 24 bit? Especially when the end result will be 16 bit and the edit will be in 32 bit float, a seemingly more compatible quotient 16 to 32 compared to 24 to 32 (??).

UPDATE: I’ve switched sites around and at this time have a new mostly proprietary method of things (previous voided). If you like my new sound and want to know more about my editing techniques feel free to ask,

Ronald Newman
http://www.SoundCloud.com/BlackDogSongs

You will probably not see this in Plot Spectrum due to the relatively course frequency resolution at low frequencies.
To “see” zero energy at zero Hz, you need to think what “zero Hz” means. “Hz” (the “frequency”) is the rate at which the audio is changing. “Zero Hz” means “not changing”, or in other words “DC”.

So “zero energy at zero Hz” means that there is no DC offset.

I’ve had problems taming low end subsonic frequencies usually caused by room roar that was heavily squelched in original '70’s rock mastering jobs that now suffer from a lack of bottom end bass that I’m attempting to put back in. Only when I do this using Audacity’s EQ I get back the subsonic sound the original mastering engineer had to severely lower.

I had to use a high pass filter set to a bottom end roll off per octave starting frequency at 1 to 5Hz and apply different db settings. What’s nice about Audacity’s High Pass filter is that it can start at 1Hz but is limited in decibel increments where other HP filters like Apples AU HPF variety allows a custom db setting. What’s not clear using either filter is the shape of the slope of roll off in how it smoothly shaves off this subsonic room roar without weakening the full sounding bass beat.

I’ve found it very difficult to surgically remove this low end hum sound without affecting the overall character of bass sounds.

Using Garageband’s compressor seems to deal with these frequencies well enough along with tweaks with its EQ but isn’t as precise as I’ld like but it does a decent job.

Audacity’s high pass filter is not limited to decibel increments. If you want, say 3.1415926536 Hz, then just type it in.

However, do note that 1 dB is a pretty small increment. Amplitude changes that are less than 1 dB are difficult to hear - in fact, many people can’t discern differences of less than 1 dB.

The shape is very close to an ideal Butterworth filter.
See: Butterworth filter - Wikipedia
and: High-Pass Filter - Audacity Manual