Hearing mic in headset while recording & setting levels

[troyinindy]

Hello all.  I am a parent of a 15 year old who is trying to basically emulate a radio station with his friends.  They are trying to set their recording levels in Audition, as well hear themselves talk when recording into Audition.  Is any of this possible?  I've tried to help them to no avail.

Recording levels- I told them to use the Windows Control Panel adjustment.

To hear themselves while recording...no success.

Anyone know?

Thanks

[ryclark]

This all really depends on the audio interface that is being used. Is the headset plugged directly into the computer? Is it USB or separate 3.5mm jacks for mic and headphone? If using the onboard soundcard then you are at the mercy of Micrsoft and Windows. What you need to be setting the playback is 'Stereo Mix' or similar in order to be able to hear playback and the incoming signal.

It is much easier if you are using an external USB audio interface that has separate mic inputs and headphone outputs rather than a combined headset.

[Bob Howes]

Ryclark is spot on--the easiest (but not cheapest) way to do this would be to get a USB interface containing "Direct Hardware Monitoring".  This will allow you to hear the mic before it even gets to the computer.  As an example, something like an Alesis i02 sells for about $100 and lets you do this.

Failing that, you can hear what Audition is recording by doing your programmes in Multitrack sessions (there are other good reasons to do this as well) and click on the little "I" symbol in the control box at the left end of the track you're recording in.   But...and it's a big but...your sounds will be doing a round trip via your computer which will introduce latency (delay) in the signal  It'll depend how sensitive your son is to this whether you can get away with it or not--but above a certain delay time it's almost impossible to keep talking if you voice comes back too far after.  You can minimise latency to a degree with the control in Edit/Preferences/Audio Hardware. 

Unfortunately, if you're using the built in sound card on your computer, you'll almost certainly be limited to what's known as MME drivers which involves your Windows in the audio path and limits how low you can go without getting glitches on your recording.  All you can do is experiment.

[Stefan Gruenwedel]

Troy: Bob Howes explains it pretty well above. You can always monitor (input) directly in Multitrack. I asked a colleague of mine who knows Audition very well and he suggests that if the latency is too long (and you’re hearing a slap-back echo) that you can lower the buffer settings (in Preferences > Audio Hardware). But like Bob said, there's a risk of possible dropouts and digital errors.

[ThomHemler]

This is utterly stupid. Who doesn't want to monitor their own voice during a record session of a voiceover?

I can't stand the echo, and dropping the sample rate results in artifacts and noise in the recording. Obviously, a DJ or VO artist did not design the monitoring option. Unusable for recording anything live.

[SteveG_AudioMasters_]

There's nothing 'stupid' about this at all - the latency is inevitable, and not caused by the software but by the computer's hardware. It takes a finite amount of time to convert a signal to its digital form and back again - which is why there will always be a small amount of latency, whatever you do. The solution, as 99.9% of users are aware, is to use straight-through monitoring through your sound device. You feed that to your headphones, and that's a direct analogue link with no delay at all. If you set your recording levels correctly, you will not overload Audition, and if by some odd chance you do, it will show up on the metering anyway.

If you have no option to do this on your system, I suggest that you purchase a better external sound device that allows it.

[ThomHemler]

Its stupid, because it could just bypass the all of the encoding, source creation, sample rate conversion, etc. and just give you a passthrough signal with 0 processing. It's not for a VO Artist's final assumption of his takes. It's a reference signal to hear tonage, inflection, emotion, and diction. It should not be a processed signal, a simple bypass output, like a digital audio board, would be perfect, and should be an option. It would be a USB or BNC cable bypass proof signal, skipping all of the processing and leaving that to the sweetener/mixer. It's stupid because there is no option for this....that's my point. It would be easy to add a simple in/out and have next to 0 delay or latency. It's so bad, the way it's set up now, that you're better off recording a VO into a $20 48k piece of software that does no processing, then do your final mixes, effects, sweetening in expensive Audition. They make these products for professional soundmixers, like myself, but forget the simple things like a bypass for RT monitoring.

[ThomHemler]

What you are referring to is a simple monitor from a sound card. Most pros have to wear many hats to stay alive and be ready for anything. It's highly unlikely any of the users, here, are Hollywood Audio Producers with Pro Tools stations, dedicated audio boards, and effects rack effects of 2000. It's a simple request, for an Adobe all-in-one set of tools called CC. I guess that's what's expected when you're a jack of all trades and master of none.

[SteveG_AudioMasters_]

And you are clearly not a master of how audio processing works in a computer, are you? It is simply not possible to provide a straight-through path through any computer without latency, even if it's a relatively low value. That's why direct monitoring is available on the vast majority of external devices, and those that don't support it aren't worth having.

The actual digitising takes place in your sound device, and the signal's then streamed out of a digital connection (usually USB) into the computer. If at this point you just looped that signal straight back to the sound device, without anything else at all happening to it, then you'd have latency. That would be down to the processing delay through the A-D and D-A processing itself. And that is why the headphone monitoring is fed from before the A-D. And there is nothing that either Adobe, nor any other software manufacturer, can do about that.

And I wouldn't make too many assumptions about what people here have access to, or do for a living either. You might be surprised...