If any can help, I'd appreciate it.

My recording desk is working well and my problem is now cleaning up the output.

Mouth noise is still a problem and I have been going through each file editting this out by hand.

By setting Audacity to waveform (db) I can easily spot them as I scroll through and they show up in the audio as a tiny spit noise.

I am willing, if not happy to do this, but I am going to feel a complete dummy if somebody on here says. 'Just set this filter to this and run it on the whole file.'

I am working with Audacity 2.3.3 on an i7 windows 10 laptop.

http://monkeyonmyshoulder.co.uk/wp/wp-content/uploads/2020/05/mouthnoise.wav


http://monkeyonmyshoulder.co.uk/wp/wp-content/uploads/2020/05/mouthnoise.png

I have diligently ignored any temptation to the EQ functions and in fact the last session required almost no tuning other than edits for spit noise and timing issues. (Big gaps / repeating the reading.)

I am still working forward rather than going back to rework the old files, as I am certain I will learn more as the project progresses.

Thanks for so much help so far in this.

ET

You could try a vinyl restoration tool that removes clicks and pops. iZotope's 'RX Advanced' would be my go-to. A less expensive RX Elements' version is available as well. In any case, set the sensitivity as high as possible without generating artifacts. Mouth noises that slip by the restoration processor can 'usually' be surgically removed using an audio "editing[" app such as Sound Forge Pro, which allows redrawing the waveform or other 'surgical' tools.

Well - if it was me, I'd simply ignore it. I expected something horrible, or troublesome, or something people would spot as a major problem and I hear perfectly normal speech from a person who probably has a tiny gap in his teeth. I don't count it as a mic placement issue, or a fault, just how some people's voices really are. Listening to just the audio doesn't cause me to highlight it as a fault, and with picture, even less so. In fact, the only thing that went through my head was his accent, and I started to try to place it, forgetting to listen for the noise. I certainly would not be looking at trying to remove it with a tool, because it's part of his natural speech, so removing it would be a task and a half, and frankly, rather pointless. Plenty of pro voiceovers have far worse natural breath sound. If you listen to Sir David Attenborough or high profile VO artists, you hear far, far worse. Stop beating yourself up for nothing.

...but I am going to feel a complete dummy if somebody on here says. 'Just set this filter to this and run it on the whole file.'

Don't know about spoken word, but this happens a fair amount with choirs and choruses. Some people make interesting noises when they talk/sing. It's what humans do. These clicks and pops usually occur way up around 9kHz or so. What editors often do in post is use a parametric EQ to make a narrow spike upward on the graph (to make the noise louder and so easier to hear and locate the correct frequency) that they can move back and forth to find the spot where the clicks/pops occur, then turn reverse the spike into a dip (to make it more difficult to hear) of some appropriate size (3-6 dB is usually sufficient).

The idea is to suppress it, not eliminate it. Trying to eliminate something like that usually leaves artifacts you can hear, if nothing else you can usually hear the "hole" you leave behind. So you just suppress it to the point that it no longer calls attention to itself.

So yeah, a small narrow dip up around 9kHz, applied to the whole file often does the trick. And often there are a couple of frequencies that need this. "Smacks" are different than "pops", and every speaker has them at somewhat different frequencies, etc. Just sayin' that YMMV.

That's a very good point about not eliminating sounds entirely.

In the parlance I am familiar with, one "attenuates" the undesirable sound rather than remove it entirely as the resulting audio 'hole' can leave it sounding not quite right. There is also the aspect of collateral damage by audio repair plugins to the tonality you wish to keep.

Andrew

My standard work flow when working with dialog is using high and low-pass filters, since there in little usable content in the human voice below 100Hz and above 10kHz. High frequency tics and such are generally in the the 8k and higher range.

It might be tedious, but fixing each individual artifact is the best sounding solution. I generally don't apply an effect to the whole track to fix short duration imperfections. Sometimes a plugin like a de-esser or de-clicker can help, but often they'll start to affect parts of the track you don't want them to.

Most likely what you're doing is the best approach, but perhaps you need a better tool. My favorite for this is clip gain automation in Pro Tools, but Reaper has a pretty good version of that.

Here's what I did on your sample in Reaper (the image is the waveform just before "decisions"):

Thanks for the input guys. I have taken a week out due to slipped disc. I think the consensus is to keep doing it by hand. Every attempt to filter it does seem to mar other places.

I've tried a few click removal plugins (on 78 & LP transfers). My experience is they need to be applied *very* carefully or they can leave low frequency artifacts.

As far as Eric's particular situation, I wonder if it would be possible to

first, split the frequency band so the offending frequencies are on one track, everything else on a second track;

next, apply some careful downward compression to the HF track, maybe around 10dB or so, and just slow enough that it would not raise gain on the mouth noises;

finally, recombine the HF and LF tracks.

As far as Eric's particular situation, I wonder if it would be possible to

first, split the frequency band so the offending frequencies are on one track, everything else on a second track;

next, apply some careful downward compression to the HF track, maybe around 10dB or so, and just slow enough that it would not raise gain on the mouth noises;

finally, recombine the HF and LF tracks.

You could simply use a multiband compressor to do that without manually splitting and recombining the frequency bands. But I don't think it would work because the noises are probably not louder than other sounds in that frequency range. I bet you'd end up with what sounds like an overly strong de-esser, which makes people sound like they're lisping.

https://youtu.be/enu-qR0H_uk

Sir David Attenborough - for many, the voice of the BBC, complete with breath noises, plosive sounds and some sibilance.

I wonder what he'd think about somebody editing out every little imperfection - his voice is quite famous, used everywhere and far from perfect.

Same with Stephen Fry and Harry Potter
https://youtu.be/phiMuhhIjAM

These have breathing noises and occasional sibilance. They could have removed these tiny imperfections, but they chose not to. Quality wise - I really hear little between the one talked about here and these big names.

You could simply use a multiband compressor to do that without manually splitting and recombining the frequency bands. But I don't think it would work because the noises are probably not louder than other sounds in that frequency range. I bet you'd end up with what sounds like an overly strong de-esser, which makes people sound like they're lisping.

I'm sure the unwanted noises are much quieter. If I were trying this (which I do not have time for) I would set the threshold just a bit higher than the level of the mouth noises. Then I'd set the time constants so there was just a small delay before the gain went back up to unity, but maybe a somewhat longer delay before downward expansion happened. With luck you might get the settings so that it took care of most of the noises automatically.

Or perhaps it would be better to use frequencies other than the mouth noise frequencies to control the expansion. So if only mouth noise were present, the "mouth noise frequency band" level would be expanded downward. If any other frequencies were present (above some threshold, obviously) then gain would be returned up to unity.

I guess any of this would depend on how similar the various mouth noises were.

Now it's starting to sound like a challenge ... but I really do not have time right now!

I certainly understand the challenge of deep diving into solving a problem like this in the midst of other things that need attending to. Reckon this thread will still be here when you get a chance to have another look at the issue.

Andrew

My standard work flow when working with dialog is using high and low-pass filters, since there in little usable content in the human voice below 100Hz and above 10kHz. High frequency tics and such are generally in the the 8k and higher range.

This has really cleaned up the sound. Thanks

End result so far. I am being less fussy about mouth noise, but rifle shooting the obvious lip smacks when it appears in a silence. Since using the clean up run filtering above from Rick, I am not adjusting any bass/ treble at all and noise reduction is almost redundant.


Greg ... I kind of follow what you are suggesting to split the tracks, but I think this is beyond my capability with Audacity as I stand today.

Please feel free to laugh, one full record session trashed due to starting too soon after eating and my stomach gurgling through 90% of it. No zips on clothes to catch on things, no curries. I am learning.

Appreciating and trying to absorb the suggestions.

Greg ... I kind of follow what you are suggesting to split the tracks, but I think this is beyond my capability with Audacity as I stand today.
I was wondering about that. I don't use Audacity, so I would not be able to give you step-by-step instructions that are applicable to your system.

Please feel free to laugh, one full record session trashed due to starting too soon after eating and my stomach gurgling through 90% of it.
Didn't you notice that right away on your headphones, before going through 90%?

I thought I had got it under control, until I came to the detailed edits and picked up so many background rumbles I decided it was easier to record the lot again than to cut and paste sections. It was the oddest thing. Even playing back the next day the rumbles actually felt like they were coming from me rather than the recording. I only realised when I replayed the same piece half a dozen times.

I am reading quite large sections at a time, usually at least an hour and repeating what I think is fluffed into the same recording. Not sure I can listen back simultaneously whilst reading and speaking. This seems to work most of the time.

The mouth lube arrived. Does improve things.

Even playing back the next day the rumbles actually felt like they were coming from me rather than the recording.
A small-scale version of "Sensurround(R)."

I'm a bit puzzled by your expression "listen back." I think of "playback" as something one does *after* recording so I wonder whether your use of the word "back" means "later." I meant to ask whether you were listening on headphones *while* recording, and whether you heard the borborygmi on your headphones then ... not upon later playback.

At least all the noises were internal, so you didn't mistake it for a neighbor practicing tuba.

Yes, a genuine sensesurround experience, if not one you'd normally pay for. Very weird.

I have not tried listening to myself simultaneous to recording.

I just configured Audacity to do this, but the effect is a weird echo a fraction after I speak. I doubt I could cope with this going at the same time as reading and speaking.

Is this the normal config for a voice recording?

If it is, I am willing to give it a bit more of a try, but first test is really mind blowing double effect. No wonder actor types are weird!

Dialog and VOs are normally recorded mono with no effects... at least on planet Earth. A boom mic and a wireless lav on separate tracks is typical (for on screen dialog) but rarely used at the same time in post

Yes, a genuine sensesurround experience, if not one you'd normally pay for. Very weird.

I have not tried listening to myself simultaneous to recording.

I just configured Audacity to do this, but the effect is a weird echo a fraction after I speak. I doubt I could cope with this going at the same time as reading and speaking.

Is this the normal config for a voice recording?

If it is, I am willing to give it a bit more of a try, but first test is really mind blowing double effect. No wonder actor types are weird!

Proper audio recording interfaces have a direct monitoring feature that allows you to listen to the live inputs with zero or virtually zero latency. With a fast computer and a proper DAW (rather than a freeware editor) you can often get quite low latency.

But would the voice over actor actually be listening to themselves as they read? I am doing around 12 hours reading and as a normal person listening to myself even in real time would be odd.

However, if this is a skill I need, I will try it.

The computer is quite fast (i7 laptop). I am also willing to buy a licensed DAW if it is required, but this is also taking me back to the question of where I started. Should I be using a stand alone recorder, rather than trying to go directly into a PC. For the budget audio book I am trying to produce I have reached a point of being content with the recording quality. I suspect my biggest limit is my skillset.

I have seen usb preamps that will take a mic and headphones, and have wondered about swapping my mic for this style of set up.

As I am about 4 hours in to recording I am loath to make changes that will affect the recording too much.

If you have a USB mic and it doesn't have direct monitoring built in the signal is forced through a round trip with several points of delay. There's the trip from the mic through its built in ADC and into the CPU, through the recording software, then through a separate DAC path in the onboard sound card. Even if better software can reduce this it can only affect the middle part of that path. That said, you can try Reaper for free. It's much more capable than Audacity. If you like it and want to own it properly it's $60. Switching software shouldn't alter your sound.

What will really allow you to monitor yourself while recording is a proper audio interface and separate mic. Although some USB mics (Yeti Pro?) do have direct monitoring, many do not. An audio interface is more than a preamp with USB connection, it's an integrated input/output solution. But it would probably mean a change in sound from your current mic. You would be able to monitor the actual sound of the mic while speaking and potentially catch problems before you invest a lot of time recording something you can't keep.

The first thing to do is determine if your mic has direct monitoring. If it does, use that instead of the computer's built in headphone output. There should be a control for balancing input and output levels. Turn any software input monitoring off.

Unfortunately for Eric, I agree with most of the above comments. Earlier in this thread, the goal was to try to identify problems, and then to make a decision about changing the gear. Now we seem to find a shortcoming in the gear: even 'though the mic might sound acceptable, it apparently lacks any zero-latency monitor output. But since you've gone ahead and started recording, changing mics mid-stream (to use half of a metaphor) will almost certainly cause a noticeable change in the sound. So I fear you are probably married to this mic at least until you get to the end of this book. I hope you can find a way to reduce the latency; even so, more than a few mSec is easy to hear.

That mic is a strange concept. I can't imagine talking into a microphone without wearing headphones.

Good summary, Greg. I came onto the forum to try to learn how to judge buying recording equipment. If I had bought, my room would still be ‘nasty’ and I would be unhappy.

I have now achieved an acceptable sound without spending anything and things I do now should be incremental improvements on what I have.

My suspicion is that whilst the central device is a laptop regardless of spec, I will be limited.

The next choice of purchase might well be back where I started which is a portable recorder, with a good quality external microphone and a zero latency headphone socket. (And headphones!) Would this be better than an integrated audio device?

Bearing in mind that at the moment, I have two books to record, I don’t have voice over aspirations. I might write other books, but I am even slower at writing than I am at recording.

However, I also think I have a vast amount to learn about technique and using what I have that will contribute to the current project without losing what I have recorded so far?

My idea is to continue through this book and learn as I am doing, then loop back and redo the start.

I owe this forum a lot of gratitude for the input you give me.

Eric,

If I were you, I'd do some research into lowering the latency of your present setup. Looking at the circuitous route described by Patrick Tracy, I wonder whether you can eliminate the parts where the [digitized] audio goes through your recording software.

Shut off the editing software, and see whether you can find a way, diddling with your laptop's sound settings, to get the mic audio to come directly back out to the headphones. That might be a shorter path, and maybe with luck you will find a way to make the latency much less than it is now.

Other than that, you have pretty well summarized the choices. From a purist perspective, the modular approach is best. A separate mic, a separate mic preamp, a separate recorder (bypassing the recorder's own preamp in favor of your external one which hopefully is better). That's also potentially the most costly approach.

Next down the line is eliminating the recorder in the above approach, replacing the preamp with one that includes a USB converter, and using your laptop for recording. The preamp/converter would have a zero-latency headphone jack.

Or, consider a USB mic with an integral zero-latency headphone jack. A lot of mics include same, so apparently most users and manufacturers recognize the need. I have one (M-Audio brand) that's at least ten years old, so clearly the concept is not new. It was reasonable quality at the time, although certainly not SOTA today. Blue makes a few similar models that do have monitor jacks, so do Audio Technica, Samson, etc. I recently saw a closeout of one sold by Marantz, who used to have a good name (but now they've slithered down into the "consumer" market). You'll find such mics listed by most music stores. My only concern is that voiceover work requires fairly high gain and low noise; a mic designed for general-purpose garage bands might not be good enough for really professional level audio books. Look for a good test & return policy.

At any rate, investigate lowering the latency of your present computer setup before you do anything else. This is something that doesn't apply to my present setup. Other people here will be able to give you more and better advice than I.

Good luck!

My suspicion is that whilst the central device is a laptop regardless of spec, I will be limited

The direct monitoring feature of a proper audio interface completely bypasses the computer. It routes the audio directly from the input of the interface to the output, without it ever going through the computer. Having an older laptop is perfectly fine. Recording one stream of audio is not a challenging task for any computer made in the last twenty years if it's in decent working order.

On some interfaces the direct monitor is a simple analog path inside the device, with a knob labelled something like Input and Computer. On some it's controlled by a driver on the computer and the audio goes through the ADC and DAC but is routed in the interface directly without going round trip to the computer. Either works, but the digital direct monitor might not work on a truly ancient laptop.

https://focusrite.com/en/usb-audio-interface/scarlett/scarlett-solo

That's true. But if you read the entire thread, you'll find that the OP is presently using a USB mic. Since he's already started recording an entire book, he needs to stick with that mic at least for the duration of this project. That being said, at the present time an all-analog path doesn't apply to him, and a separate audio interface doesn't apply to him. So the immediate question is how to get him very low latency monitoring while he's recording himself.

I've read articles saying 8 - 10mSec is acceptable latency, in terms of musicians staying in sync. But from my own experience, I've found that less latency than that can still be disconcerting, because my voice coming through the headphones is phase shifted from what I hear via bone conduction. One's voice can in some cases sound quite strange. It's hard not to try to compensate for that in one's delivery.

That's true. But if you read the entire thread, you'll find that the OP is presently using a USB mic. Since he's already started recording an entire book, he needs to stick with that mic at least for the duration of this project.

I know, I'm just trying to get him to understand the issues. For a single stream it might be possible to get latency quite low, but using separate input and output devices is going to work against that.

Setting buffers to lower values is probably the first step. Lower it until dropouts occur then go back up one step.

Also, using ASIO drivers would help, if available. Audacity can do that but it's something you kind of have to be a coder to enable. An actual DAW would be better for that.

I wonder whether anybody makes a USB audio monitor. Not a playback device. A device with a USB pass-through that would (1.) accept data from a USB mic, (2.) tap it and convert it to analog with as little latency as possible, and (3.) pass it through to a computer for recording.

I might google it, but I'll be surprised if I find one. It would just be a "band-aid" for doing things in a less than ideal way in the first place. I hasten to point out that the OP would be fine right now if his present mic had an analog monitor output. How sad that none of us (including me and the OP) caught this before he started recording.

I doubt it.

The point of a pass-through for monitoring is that it bypasses the latency that results from digital data handling and/or processing. To achieve that it's done on an analogue basis. A USB dongle that converts the audio data back to something you can hear has to run it back through a (digital-audio-converter) DAC to function and there is always a delay time for this to happen.

The headphone out facility on the Rode Podcaster (http://www.rode.com/microphones/podcaster) (for example) gives zero latency, and you'd be making a safe bet that it's all done at the analogue stage within the mic before conversion to digital data to be sent over USB.

Andrew

Of course. Once the signal out of the mic is USB, one A>D is involved. A monitor box (if such existed) would need to have one D>A. Still, it's possible that the total latency from those two devices might be less than the latency when feeding the USB mic through a computer (and back out again). It would depend on the latency in the given computer, as well as the speed of the D>A in the monitor.

As I expected, a quick search did not find anything like this.

A lot of newer interfaces do direct monitoring after the ADC, but it stays in the unit where the circuitry is optimized for low latency.

If the OP has a Yeti, perhaps upgrading to the Yeti Pro would retain the sound he has now.

How sad that none of us (including me and the OP) caught this before he started recording.

Please, no self recriminations for the flood of great advice. You had no concept of how rookie I am!

I am learning as we go. I never had a monitor facility, and reading a couple of voice over blogs, they suggest a ‘real artist’ would prefer to hear his own voice than the playback … guess that means I am a real artist.

Staying in perspective, I doubt I will ever seek much voice over after my books are recorded. I am now 75% through the first book. With the help of the forum, I am creating a good quality recording.

I would never have known I could monitor real time if I hadn’t been amused by the stomach grumbles. Fixed by controlling my diet! Plus, I can now check the monitor to confirm all sounds well before starting to record, which is another step.

Obviously, the geek in me wants to understand/ learn and possibly acquire better equipment.
I did check the PC and windows 10 allows listening to an input device, whilst this has less latency at least cutting any app software out of the loop, it still causes me to speak much slower and to grind to a stumbling halt. Really odd effect. I also tried the headphone jack socket, albeit with a phone headset instead of USB. This appeared to be the lowest latency, but still too much for my brain to cope.

I ‘could’ acquire the more expensive usb mic, but I would also need a reasonable set of headphones to go with it. Would this be putting lipstick on a pig?

Taking the comment that discrete might be better.

The cost of an audio input / headphones / mic are not actually much more than the yeti pro/ headphones.

Or, back to original concept, headphones, mic and the TASCAM DR 40X (or something similar) gives me a stand-alone recording device. And again, is not much different in price? This device supports a quality external mic and has a headphone jack. (Will confirm if this is a monitor port)

If you dig around you may find a used interface at a good price. I recently saw a Lexicon Alpha sold for ~ $25. I also saw new / discontinued Marantz USB mics, which have headphone jacks, being sold for $40. Are these things better, or worse, than what you now have? Ideally when shopping for low price you want a try / return option so that narrows down the field.

All standalone audio recorder I've seen have had headphone jacks which allow you to monitor while recording. I own three different Tascam models, at least one Olympus, one Marantz, plus a few pocket size Sony and Philips "voice recorders." All of these have 3.5mm headphone jacks. At some point I've tested all of them by recording my own voice. I'm sure I would have noticed any latency.

I've observed that a delay of ~150mSec will make most people slow down, stutter, and finally just stop in befuddlement. When "breaking in" new announcers, we used to secretly patch their headphones to the signal coming from the confidence (playback) head on a 3-head R-R tape machine so they'd hear their own mic with intentional delay. My advice is not to do that to anyone unless you can run faster than they can.

Be careful with older Lexicons - I have an omega and a colleague had an Alpha - they're wonderful, but check the driver situation. I don't think there is a Windows 10 version, and it's unsupported now.

Greg speaks the truth.

I was doing a news read for radio back many years ago and my headphones had somehow been switched to receiving the off-air receiver monitoring feed instead of the studio audio panel. The delay between the studio and coming back from the transmitter so messed me up, and I dare not ponder what it sounded like to the listeners. In hindsight I should have tossed the headphones off my head and over my back and continued but, nay, I battled on to the very end.

For the uninitiated, having the off-air (radio receiver) source is so you know immediately if the transmitter has gone down or up in flames and you are now only talking to yourself.

Andrew

PS. Then there was the time I was doing the Saturday brekky shift (announcing) and the last announcer from the night before (overnight was on automation) must have surely been going deaf. The moment I switched on the mic to do my first voice announcement (about 6:30 am or slightly earlier) the sound levels coming from the headphones were actually enough to be picked up by the microphone and gave me a classic (and very loud) 'PA feedback' squeal ... in stereo. This insanely loud squeal only started when I commenced speaking. The audience will have heard it at the very moment I did, in addition to what my "good morning ..." rapidly devolved into. Never did quite get to the bottom of that one but at least I was very awake from that moment onward.

Final addendum (Probably)

http://monkeyonmyshoulder.co.uk/wp/wp-content/uploads/2020/06/newsetup.wav

Couldn't not do it. I had never tried/ experienced using a monitoring setup. This is so much better.

Finally went with the tascam (it had to happen) an AT2020 condenser mic and studio spares own closed headphones. Combined with the rest of the changes, this is so sweet.

I now get how this works ... how did I go for so long without monitoring as I recorded?

Decision is still to press on and fully record this book and then loop back and rerecord the earlier files on the basis that I will keep learning.

Thanks for the guidance.