Firewire based mixer recording into DP, level settings?

[Gibble]

Hello all.

Well, my equipment finally came in (just over 2,000$ worth of gear). So now I have a couple of questions regarding setting the levels in the recording chain.

The recording environment is an auditorium with various performers throughout the recital. (vocal soloists with piano accompaniment, acoustic guitar-both solo and duets, and violin soloists with piano accompaniment)

The signal chain consists of a MS pair with a third HyperCardioid Mic for additional focus on the vocalist (These three mics are placed coincident to avoid phasing problems and are located about 10 feet away from the stage). There are also two cardioid condensors placed on stage for picking up acoustic guitar performances

1) The mixer sends each channel, POST fader, to the DAW (DP) via firewire.

2) Each track is being recorded live into the DAW for processing later.

3) At the analog side of the chain (mixer) should the channels all be set to unity or under to avoid clipping?

4) At the DAW side of the chain (inside the computer) what should the levels of each recording track be set at?

5) I would imagine that the use of a compressor or limiter plugin should be used to prevent signal clipping...what should the meters (in the DAW) be peaking at, (say -12db or unity?) post plugin, so that the signals are at a good level prior to post recital processing.

(Okay, you can tell I'm a complete rookie at this, but what better way to learn than getting advice from some of you pros.....thanks in advance)

Gibble

[HCMarkus]

The analog part of your chain is the most critical, assuming you record at 24 bits in DP.

Every piece of gear has its own sweet spot... your assignment, should you choose to accept it, is to find the spot for your gear. It's always noise vs distortion/headroom. Use care in gain-staging right to your AD converters and record with enough headroom so that an enthusiatic performer doesn't overload a mic pre (watch you mixer's meters and use your ears) or hit full-scale at your converter (as confirmed by your converter and/or DP's metering). For every bit beyond 16, you get 6 dB headroom, so don't worry about having to get real close to digital full-scale for a clean recording, 'cause almost everything ends up being dithered down to 16 bits for CD anyway.

Adding a limiter plug in will not stop clipping! You should not need an analog limiter/compressor to record a recital. If the signal makes it thru your converter without clipping, DP will capture the audio perfectly. Once audio is in DP, it floats around in DP's 32-bit environment until it hits a physical output.

Depending on how live the hall is, you might want to push your soloist spot mic a tad closer for enhanced presence. You can always delay/advance a channel in DP if its mic isn't physically lined up with other mics. Have fun, but do a practice run, i.e. record a rehearsal, before the real show!

[Gibble]

Thanks HCMarkus

Adding a limiter plug in will not stop clipping! You should not need an analog limiter/compressor to record a recital. If the signal makes it thru your converter without clipping, DP will capture the audio perfectly. Once audio is in DP, it floats around in DP's 32-bit environment until it hits a physical output.

Okay, I understand what you've said here, but, what do you do about the occasional peak in the performance that would cause your carefully set levels to go over unity to the point of causing distortion? Do you set the analog levels (at the mixer) low so that peaks don't push over 0db?

Thanks again

Gibble

[HCMarkus]

If extreme peaks are a problem, you can use an analog compressor/limiter on each mic channel, but I would suggest just setting a slightly lower input level at your Mic Pre or, if your Pre doesn't clip too hard, at the fader. If your signal path is clean, noise should not be a problem. Remember, your signal has to hit a Mic Pre before you could insert a compressor/limiter anyway, and the Mic Pre stage is where you will generate the most noise. While using limiters can protect your AD from overloads, other issues arise, in particuar compression artifacts and image-shifting when limiters don't track perfectly from channel to channel, not to mention the set up time. When it's time to mix, you can use a plug in to finesse your dynamic range on a song per song basis. Also, you can cut or fade extraneous noise that may be audible between songs but will be masked when the music is playing.

[Gibble]

If extreme peaks are a problem, you can use an analog compressor/limiter on each mic channel, but I would suggest just setting a slightly lower input level at your Mic Pre or, if your Pre doesn't clip too hard, at the fader. If your signal path is clean, noise should not be a problem. Remember, your signal has to hit a Mic Pre before you could insert a compressor/limiter anyway, and the Mic Pre stage is where you will generate the most noise. While using limiters can protect your AD from overloads, other issues arise, in particuar compression artifacts and image-shifting when limiters don't track perfectly from channel to channel, not to mention the set up time. When it's time to mix, you can use a plug in to finesse your dynamic range on a song per song basis. Also, you can cut or fade extraneous noise that may be audible between songs but will be masked when the music is playing.

If I understand what you're saying here correctly, then the old analog approach of recording the signals as hot as possible no longer applies when working in the digital domain. If this is the case is there a level that the sound should be recorded at (-12 db for example)?

Thanks for your help HCMarkus

Tera

[HCMarkus]

If I understand what you're saying here correctly, then the old analog approach of recording the signals as hot as possible no longer applies when working in the digital domain. If this is the case is there a level that the sound should be recorded at (-12 db for example)?

Analog tape has a very cool thing it does when it is hit hard: a nice soft rounding of transients and a very musical compression. That's why a lot of folks still track to tape (especially drums) and transfer to a DAW for editing and mixing. On the other hand, digital overs of any duration beyond a few samples are like sandpaper in your Skilsaw, with blood and carnage everywhere. It's actually a great effect, but not particularly recital-friendly.

In the digital realm, record only as close as you can get to full code without risking overload... and then leave 3-6 dB extra headroom just to be safe. The nature of the signal you are recording will impact your decision on levels: if you are recording an unvarying sound, like a sine wave (how exciting!), you can better predict what is going to happen than when you are recording a group of 12 year old singers on Ritalin. Again, 24 bit provides 48 dB in headroom over 16 bit CD audio, which is what your recordings are almost certainly destined for, so err on the side of caution in levels you send to your AD.

I guess -12 dB sounds pretty good.

Technical PS: it's dB, not db, in honor of his audio highness Alexander Graham Bell.
http://en.wikipedia.org/wiki/Decibel#History

[Gibble]

Thanks for all the great information.....it's helped out a lot!

On the other hand, digital overs of any duration beyond a few samples are like sandpaper in your Skilsaw, with blood and carnage everywhere. It's actually a great effect, but not particularly recital-friendly.

*Chuckle*, vivid description!

In the digital realm, record only as close as you can get to full code without risking overload... and then leave 3-6 dB extra headroom just to be safe.

Okay, so depending on the situation the signal level in may actually be quite a bit lower than unity (digital domain).

....recording a group of 12 year old singers on Ritalin. ...

Ahhh, I see you've attended some of the same recitals .....

Technical PS: it's dB, not db, in honor of his audio highness Alexander Graham Bell.
http://en.wikipedia.org/wiki/Decibel#History

Thanks for dB information.

I really appreciate your help HCMarkus

Gibble (Tera)

(I don't generally let my name out, it must have slipped out on the previous post.....late night)

[Shooshie]

Digital music and digital photography share a similar phenomenon in comparison to their analog counterparts. In analog photography you shoot for the shadows and develop to compress the highlights. In digital it's the other way around: you shoot for the highlights and compress (or expand) for the shadows.

Put simply, it's because on analog film if it's too dark, there simply isn't any information there. So, you be sure to meter for your shadows to make sure your information is there. Then the highlights can be brought under control by compressing. The info is there in those highlights; it just needs to be wrangled down to a manageable size.

In digital, it's the other way around. You can always brighten those shadows if they aren't TOO abused. But highlights? Once you've blown your highlights, there's no information left in them. So, you make sure you don't overdo your highlights, and you adjust the picture to bring up those shadows.

In audio, analog can handle overages very nicely with only a little bit of compression, but the low end gets lost in noise. Most of your efforts are focused on getting the quiet spots separated from the noise with enough detail to make them discernable. The peaks can be handled pretty easily if you just take care of the low end vs. the noise, and you generally do that by raising the gain; recording a little hotter.

In digital audio, you have to aim to keep the peaks from losing their detail and information. Once you blow them out, there simply is no way to get the information back unless you're recording in 32 bit or 64 bit -- something very rare in this world. Think "highlights" in digital pictures; the information has exceeded the digital capacity to record it. But at the other end of the scale, the noise is so low that you get great definition of even very quiet sounds, so -- like the shadows in digital pictures -- you can generally compress those upward if needed since the detail is there. So, we take great pains to prevent our levels from exceeding unity or at least a few dB above it -- whatever is the physical limit of our software. We prefer to err on the quiet side, because with the full picture just moved down a few notches, we can always bring it up and/or compress it to a level that fills those monitors up with fully-detailed sound. Err on the loud side, and you've just got no picture.

To summarize:

In the analog world, you focus on the quiet spots; shadows and noise. Then in the mix (or printing) you bring down the highlights and peaks until they're manageable.
In the digital world, you focus on the loud spots: highlights and peaks. Go beyond the limit, and you have no usable information. Then in the mix (or printing) you wrangle the shadows and quiet spots until they are at an acceptable level for your perception.

And a little more info:
A "compander" like Waves' C4 processor or their Linear Phase Multi-Band Processor makes this a very easy and musical process over which you have a great deal of control, while using a limiter like the L2 Ultramaximizer to act as a brick wall against further peaks that might develop from processing.

I hope I've clarified it. If you're not into photography, this might not make as much sense. But I think the picture is pretty easy to see with a little imagination.

A belated welcome to the forum. Feel free to ask any question that will help you understand what you need.

Shooshie

[sdemott]

A couple of points to consider:

Compression is something to be very careful with in classical music. It is usually not used in the recording chain and only very lightly with a very high-end & transparent unit (not plugin) during the final mixing (if at all).

As far as mixer levels. Figure out how the Hw is configured. The AES standard is that -20dBfs = 0VU = +4dBu, the EBU standard is -18dBfs = 0VU = +4dBu. So calibrate your equipment appropriately.

More or less you want an analog "0" reading to hit your DAW at either -20dBfs or -18dBfs.

From there - just keep the levels between reference (-20/-18 dBfs) and about -9dBfs and you'll be fine. More than enough headroom and no need to worry about compressors or limiters for peak management.

HTH

[Gibble]

To summarize:

In the analog world, you focus on the quiet spots; shadows and noise. Then in the mix (or printing) you bring down the highlights and peaks until they're manageable.
In the digital world, you focus on the loud spots: highlights and peaks. Go beyond the limit, and you have no usable information. Then in the mix (or printing) you wrangle the shadows and quiet spots until they are at an acceptable level for your perception.

And a little more info:
A "compander" like Waves' C4 processor or their Linear Phase Multi-Band Processor makes this a very easy and musical process over which you have a great deal of control, while using a limiter like the L2 Ultramaximizer to act as a brick wall against further peaks that might develop from processing.

I hope I've clarified it. If you're not into photography, this might not make as much sense. But I think the picture is pretty easy to see with a little imagination.

A belated welcome to the forum. Feel free to ask any question that will help you understand what you need.

Shooshie

Thanks Shooshie, I had to read through that a couple of times but it makes perfect sense.

Now that all of the equipment is here the next step is to start experimenting with levels and microphone placement.

I've already set the gear up to test out the connectivity with the computer and DP (seems to be working okay, no problems with either the drivers or any weird behaviors).

One initial observation is that the mic-pres on the mixer might be a little on the low power side until you reach the last quarter of the gain then it jumps up quite rapidly. (is this normal for mic-pres on mixers?) I'm wondering if it might be wise to go to an outboard pre-amp.

Gibble