Clipping/peaking out

[rob aylestone]

It's has been done in studios for a very long time, but time and techniques do change. The good thing is that we can all continue our old tried and tested processes. However, for newcomers, the old advice could simply be less sensible now. How many of us have not been guilty of overcooking the compression and being stuck with it when we do a remix, or thinning out? We all overdid it when learning. Now newcomers just don't have the arms twisted behind our backs that we had back then. Non-destructive techniques should be promoted. As an example, so many people use tape saturation plug-ins as just another effect. When tape was involved you often had little choice if you needed less hiss. I know everyone loves tape saturation, but I did not. Things move on so fast.

 

[ecc83]

Whilst there are some folks who cannot get on with DAWs I do not think they are SO problematic that they should be avoided altogether. I do not count myself as any kind of computer whizz but I have used several DAWs over the years. Cubase, Sonar/Cakewalk, Adobe Audition (1.5 and my only bit of cracked software, a gift with a computer for my 60th ***K! That was 17 years ago) Audacity and my main software Samplitude Pro X 6. Yes, I get the odd setup problem but then Emma is having HARDWARE problems! All this boils down to is some hard study and a systematic approach. I only do REALLY basic stiff with Sam 6.

The very reasonable cost of Reaper buys you functions that would cost you many thousands to get in hardware .

Yes Rob, many engineers in the classical field only used about half the noise reduction possible with Dolby A. they used the rest to record at a lower level and get far cleaner recordings. The other big advantage of DA was of course the ability to make copies with little noise build up...PROVIDED of course one was scrupulous in level matching throughout the frequency range.

Dave.

 

[drtechno]

Hi guys,

I wondered if anyone has any tips on setting the gain appropriately for vocals? I find that when I belt out louder notes or choruses, I get a lot of clipping but when I turn the gain down (even slightly), the vocal is too quiet to work with. I've experimented with compressing, different mics, mic technique, preamp settings etc but I just can't seem to find the sweet spot.

My setup is a condenser mic into a Voicetone T1 into an Art Studio preamp into a Tascam DP008EX. To make matters more confusing, there are separate gain settings on the Voicetone, preamp and portastudio, and the preamp will often indicate no clipping but the meter on the portastudio itself will be peaking out so I don't know which should take precedence.

Please help!

It doesn't help the tascam is geared towards low signal level. I would put a Shure A15AS attenuator right before you go into the tascam and pad it -25db so you can track at -10dbu nominally.

 

[ecc83]

Note of caution about that Shure attenuator. It is listed for use with microphones AND line signals. I have checked the specification and it has an input resistance of only 1000 Ohms and that is too low for many AIs, especially those of the budget persuasion. Many interfaces and other devices, synths come to mind, can use an output device that is not happy loaded with less then around 3K Ohms and will run into distortion with a 1K load.

So, fine for microphones but not so good for line signals which really should not be loaded below the common standard of 10k Ohms.

All that said? Some while ago I searched for line attenuators and found none. Many suppliers do not even tell you what the input and output resistances are!

 

[BeGrody]

It doesn't help the tascam is geared towards low signal level. I would put a Shure A15AS attenuator right before you go into the tascam and pad it -25db so you can track at -10dbu nominally.

What makes you say the Tascam is geared towards "low signal level." The Tascam has multiple input settings as well as a switch on the back for selecting various input levels which includes line level.

 

[BeGrody]

Whilst there are some folks who cannot get on with DAWs I do not think they are SO problematic that they should be avoided altogether. I do not count myself as any kind of computer whizz but I have used several DAWs over the years.

Hi ecc83. Just want to offer another perspective. I am a software engineer by trade and I suppose that qualifies me as a "computer whizz." I hold a computer science degree and a masters degree in software engineering and have about 20 years experience in the field. You have most likely used my software because my area is highly-public "web scale" web services and applications.

Further, I began using DAWs long ago and have used Logic, ProTools, Cubase, Ableton, CakeWalk, FruityLoops, etc. for both creating music, and capturing others music, and mixing with these pieces of software. I've also done voiceover, film, and speech work. All told, I can't even remember how many different pieces of in-the-box audio software I have used. I have even used all the auxiliary products like EZDrummer, Reason, etc. I've used them live on stage sometimes.

Now, that doesn't qualify me as anything special, and that's not my point.

My point is that this has given me the perspective that DAW's and software are not worth it sometimes. Even with all that experience and the ability to do complex stuff with software, I personally have found that such technology constantly works against the creative process and hasn't been constructed in a way that enables a smooth "flow." I have heard highly-visible mix engineers say the same. Chris Lord-Alge gave an interview to TapeOp about how he is not going to abandon his mixing board or his rack of analog gear just because the same is "possible" in-the-box. Although what's "possible" in software is the same as hardware, the actual experience is often quite degraded.

All that being said, I do not imagine to even recommend one way or another to anyone else. But I know what I have experienced and love to share it as a point of reference.

YMMV

 

[drtechno]

What makes you say the Tascam is geared towards "low signal level." The Tascam has multiple input settings as well as a switch on the back for selecting various input levels which includes line level.

Because all of them have been like that. Regardless of model. And it has to do with their selection of parts and source voltages they choose to engineer around.

Please remain on topic here. There is a proper place for introductions, and this thread is not the place.

 

[BeGrody]

Because all of them have been like that. Regardless of model. And it has to do with their selection of parts and source voltages they choose to engineer around.

I am not sure, you may have in mind the Tascam cassette-based recorders. This is a digital recorder with the following specs:

• XLR [BALANCED]
  Connector: XLR-3-31 (1: GND, 2: HOT, 3: COLD)
  Input impedance: 2.4 kΩ
  Standard input level: −8 dBu
  Maximum input level: +8 dBu
  
  
• Standard jack [UNBALANCED] Connector: 6.3mm (1/4”) TS standard jack
  Input impedance: 10 kΩ or more
  (when INPUT A switch set to MIC/LINE) 1 MΩ
  (when INPUT A switch set to GUITAR)
  Standard input level: −10 dBV
  Maximum input level: +6 dBV
  Headroom: 16 dB

Maybe you consider this "low level" but I wouldn't. It has 16dB headroom and maximum input level at +8 dBu. You suggested trying to run -10 into it.

 

[drtechno]

Headroom: 16 dB

Maybe you consider this "low level" but I wouldn't. It has 16dB headroom and maximum input level at +8 dBu. You suggested trying to run -10 into it.

Because 18 to 22 db headroom is the pro recording standard. Even though this is not pro audio device. Attenuation of a high level signal will also allow the operator to gain stage the noisefloor out of recording as well. If you can't contribute a logical awnser to the OP problem, may I suggest you post engineering questions in a separate thread.

 

[BeGrody]

While true, that doesn't make too much sense in the context of this specific question. The gear OP has is set up to accept input level between -10 and +6 so any level in this range is acceptable. The only reason to use an attenuator in addition to everything else in that chain is if there is no way with her equipment (the ART Tube MP and the Voicetone pedal) to get the level into this range before coming into the Portastudio.

 

[drtechno]

While true, that doesn't make too much sense in the context of this specific question. The gear OP has is set up to accept input level between -10 and +6 so any level in this range is acceptable. The only reason to use an attenuator in addition to everything else in that chain is if there is no way with her equipment (the ART Tube MP and the Voicetone pedal) to get the level into this range before coming into the Portastudio.

Lots you have to learn about gain staging. That is why you need to ask this question in a new thread.
To achieve maximum signal to noise in all devices before the recorder, the signal level would be too great, and that is why the operator would attunate the signal before going into the low voltage recording device.
That is why I keep a pair of these attenuates in my live kit in case someone wants to hook their recorder up.
Also, if you noticed I suggested a switchable instead of a fixed because this is not a one shoe fits all, when different devices are connected in the recording chain.

 

[Farview]

I am not sure, you may have in mind the Tascam cassette-based recorders. This is a digital recorder with the following specs:

• XLR [BALANCED]
  Connector: XLR-3-31 (1: GND, 2: HOT, 3: COLD)
  Input impedance: 2.4 kΩ
  Standard input level: −8 dBu
  Maximum input level: +8 dBu
  
  
• Standard jack [UNBALANCED] Connector: 6.3mm (1/4”) TS standard jack
  Input impedance: 10 kΩ or more
  (when INPUT A switch set to MIC/LINE) 1 MΩ
  (when INPUT A switch set to GUITAR)
  Standard input level: −10 dBV
  Maximum input level: +6 dBV
  Headroom: 16 dB

Maybe you consider this "low level" but I wouldn't. It has 16dB headroom and maximum input level at +8 dBu. You suggested trying to run -10 into it.

Right. Pro level,the level that the preamp is probably kicking out, is +4 with a maximum level of +20. Those inputs are consumer level.

Either way, nominal level on the preamp only gives you 4db of headroom on that input. (Thus the clipping)

 

[Mickster]

Totally off topic....but....I have to say.....having followed this thead and taken time to absorb and research......I've really learned a lot from it. Really. Thanks everyone.

Mick

 

[ecc83]

"That is why I keep a pair of these attenuates in my live kit in case someone wants to hook their recorder up."
Good stuff ^ Yeah, 'kids' today don't know they are born when it comes to signal exchanging.

40 odd years ago when I was in 'sound re-inforcement, am-drams, AGMs and such. I had an Aluminium box that had just about every audio* connector known to man (in UK anyway) RCA, DINs boith A and B and a six pin, XLR jacks, A&B (PO) 1/4" and two other sizes. These wired down internally to 'choc-block' terminals so that signals could be routed correctly since not all equipment makers followed the same connection regime. DIN especially was a nightmare because neither the Americans nor the Japanese really understood it, not the pin outs nor the levels!
Also a 100k stereo log pot in a tin to chop levels where required since I never knew what kit someone would rock up with wanting a feed.
One especially useful tin had 2 RCAs in and out and turned a nominally 1 V 'flat' signal into about 5mV corrected to vinyl RIAA response. Thus the mag pup input on a hi fi amp or tape machine could be pressed into service as a line level device.

The system used by pro studios is that just about everything works at "unity gain" and by far the most common level now is +4dBu or about one volt rms (usually with at least 20dB of headroom) Twenty years ago it would have made 'domestic' audio much more expensive to do that. Now, not so much but gear still abounds with no logic or sense to its operating level.

*And, for a long time we all had to contend with EIGHT different mains plugs!

Dave.

 

[BeGrody]

Lots you have to learn about gain staging. That is why you need to ask this question in a new thread.
To achieve maximum signal to noise in all devices before the recorder, the signal level would be too great, and that is why the operator would attunate the signal before going into the low voltage recording device.
That is why I keep a pair of these attenuates in my live kit in case someone wants to hook their recorder up.
Also, if you noticed I suggested a switchable instead of a fixed because this is not a one shoe fits all, when different devices are connected in the recording chain.

"Lots you have to learn about gain staging."

I don't think so. What I've said is true or you can correct it. Ad hominem fallacy right here.

 

[BeGrody]

Right. Pro level,the level that the preamp is probably kicking out, is +4 with a maximum level of +20. Those inputs are consumer level.

Either way, nominal level on the preamp only gives you 4db of headroom on that input. (Thus the clipping)

You're both literally just speaking in generalizations as though they are 100% right all the time. The ART Tube MP has a variable output level that is max +10.

This is a direct copy/paste from the manual downloadable on the ART site:

The output control sets the output level of the TUBE MP. When the control is fully counterclockwise the output level of the TUBE MP is zero. Turning the control clockwise increases the level of the output to a maximum of +10dB of gain. This gain is in addition to the existing input gain.

There's no attenuator necessary here. There's an output knob right on the preamp which in addition to the input gain knob can output less signal than the input on the Portastudio. All OP would have to do is turn the knobs counter-clockwise a bit. Not buy an attenuator.

 

[BeGrody]

I get a lot of clipping but when I turn the gain down (even slightly), the vocal is too quiet to work with....and the preamp will often indicate no clipping but the meter on the portastudio itself will be peaking out so I don't know which should take precedence.

This is the original problem from the OP. It's probably as simple as she needs to turn the output knob on the ART all the way counter-clockwise (because it is on top of input gain on the unit) and then adjust the Portastudio down until it just barely isn't clipping anymore. (After making sure she's set the Portastudio to a line-level input setting.)

But many in this thread are jumping in with the "ya gotta buy more stuff" answer, whether it's an audio interface and a DAW, or an attenuator, or whatever.

 

[JamEZmusic]

This is the original problem from the OP. It's probably as simple as she needs to turn the output knob on the ART all the way counter-clockwise (because it is on top of input gain on the unit) and then adjust the Portastudio down until it just barely isn't clipping anymore. (After making sure she's set the Portastudio to a line-level input setting.)

But many in this thread are jumping in with the "ya gotta buy more stuff" answer, whether it's an audio interface and a DAW, or an attenuator, or whatever.

Have you got any experience with the ARTtubeV3?

I have it but I'm not so sure I'm getting the most from it. Are you looking at the gain reduction meter and slamming it? Or are you looking to just make it fluctuate slightly? Let's say you're recording acoustic guitar (strumming) is the unit worth using for something so dynamic when I have no other means of dynamic control going into the ArtTube?

It sounds great on bass guitar, this is probably what it's most known for.

I've been re-amping out through my DAW back into the ARTtube, then back into the LineIn on my audio interface so I can have a compressed signal going out of my DAW into the ArtTube, then also I don't care if I overdo it on the re-amped signal because I still have the original copy. It's my go-to method of using it because there is literally zero chance of messing up doing it this way. If all goes well I can just mute the original, at worst I'll pinch the re-amped signal and use it for low end or low mids

 

[BeGrody]

Have you got any experience with the ARTtubeV3?

I have it but I'm not so sure I'm getting the most from it. Are you looking at the gain reduction meter and slamming it? Or are you looking to just make it fluctuate slightly? Let's say you're recording acoustic guitar (strumming) is the unit worth using for something so dynamic when I have no other means of dynamic control going into the ArtTube?

It sounds great on bass guitar, this is probably what it's most known for.

I've been re-amping out through my DAW back into the ARTtube, then back into the LineIn on my audio interface so I can have a compressed signal going out of my DAW into the ArtTube, then also I don't care if I overdo it on the re-amped signal because I still have the original copy. It's my go-to method of using it because there is literally zero chance of messing up doing it this way. If all goes well I can just mute the original, at worst I'll pinch the re-amped signal and use it for low end or low mids

I have experience with it.

I'm not sure I'm following the questions in the second paragraph. It's a mic preamp. You can record anything with it. Acoustic guitar, sure. Personally, I would say it is context-dependent and outcome-dependent regarding whether I'm "slamming it" -- it's more about the sound I'm getting rather than what the meter is doing.

But you wouldn't use it specifically for controlling dynamics. (Unless you have the MP/C which is the model with a compressor?) Instead, you would record whatever it is that you're recording with plenty of headroom and then compress it during mixing with a compressor.

What makes you say that about it being most known for bass guitar?

Since it's a mic preamp, the "correct" input source is a microphone. But that's fine if you use it in a different way. If you get what you want out of it, then that's great.