Let's continue the conversation about "gain structure"

[lifelyrics]

Hi Mike,

I'm pasting part of your comment here, so we can hopefully talk about it.

gecko zzed wrote:
Incidentally you refer to dynamic mikes and overloading the DP004. I expect this is not a problem with the mikes, but more a problem with gain structure. 'Gain structure' may not mean much to you now (forgive me if I'm wrong, I really have no knowledge of your technical prowess and I could be wrong), but it's worth making this topic one of your next steps. (If you want to read his entire comment, you can find it @ recording techniques -> Home Recording's Dirty Little Secret -> last page, I believe)

No offense taken. First off, I'd like to say, I love your feedback. You are always informative and insightful.

My understanding of "gain" is simplistic, I'm sure, because I have only used it when I worked in Audacity, and now in Sonar HS 7XL (purchased a month ago - after the recordings for the memorial service). In my terms, it is comparative to whatever else is going on in the mix + or -, therefore I wouldn't even think to use it on a single track, unless the whole thing was too quiet. In audacity, when I ended up with a wav file that was too small for me to see what was going on, I just used "amplify", and then worked from there. I would then adjust gain/pan, etc. to get the right blend with all of my tracks together.

Looking back on it, Hubby thought part of the problem was that we used a low->high impedance converter to go from the XLR to the 1/4" in on our DP, because we were in a time crunch, and had to go with what we had. Would that be a possible reason?

We hadn't even really had time to do much experimenting with the DP before being thrown into the project, since we just got it after Christmas. Then to complicate things, as I said somewhere else, my friend wanted reverb, which isn't in audacity (well, it says it has the effect, but it was unacceptable.) So, we borrowed the church's Mackie mixer (don't remember which one - I think 12 channel), and ran through it to DP. We adjusted gain on the mixer, but had to have everything set so low, I ended up with teeny tiny wav files, which I "amplified" so that I could work with them. I think it's because the mixer acted as a pre-amp, and made the signal too strong for the DP. Is that correct?

So, now we have a proper mic cable if the situation ever arises again, that we need to do vocal tracks on the DP (XLR -> 1/4"). That is how we conducted my mic test the other day - straight to DP, and I found that the input level could be set 1/2 way instead of barely past zero.

This week, my new firewire interface, condenser mic, pop filter, and studio headphones will be arriving, and I will get into that phase of my education - putting all of it together with my SONAR software and my laptop. I'm excited. I love learning. I welcome any additional instruction/information you feel like offering!

 

[Farview]

My understanding of "gain" is simplistic, I'm sure, because I have only used it when I worked in Audacity, and now in Sonar HS 7XL (purchased a month ago - after the recordings for the memorial service). In my terms, it is comparative to whatever else is going on in the mix + or -, therefore I wouldn't even think to use it on a single track, unless the whole thing was too quiet.

Your confusion comes from the use of the term gain in several different contexts. It means slightly different things depends what you are doing.

Every pice of equipment you run a signal through is going to affect the level of the signal you are sending it. The trick is to take the mic signal and add gain to it to bring it up to line level. Then the idea is to keep the signal at line level all the way through the signal path. The 'gain' control just adjusts the signal level up or down.

In audacity, when I ended up with a wav file that was too small for me to see what was going on, I just used "amplify", and then worked from there. I would then adjust gain/pan, etc. to get the right blend with all of my tracks together.

There is a zoom control that will let you see the low level stuff without actually affecting the volume of the track. There is no reason to change the gain so you can see it better?

Looking back on it, Hubby thought part of the problem was that we used a low->high impedance converter to go from the XLR to the 1/4" in on our DP, because we were in a time crunch, and had to go with what we had. Would that be a possible reason?

Nope, that's exactly what you need to use in order to get the mic into a high impedance jack. Is there no xlr input on your preamp/interface?

We hadn't even really had time to do much experimenting with the DP before being thrown into the project, since we just got it after Christmas. Then to complicate things, as I said somewhere else, my friend wanted reverb, which isn't in audacity (well, it says it has the effect, but it was unacceptable.) So, we borrowed the church's Mackie mixer (don't remember which one - I think 12 channel), and ran through it to DP. We adjusted gain on the mixer, but had to have everything set so low, I ended up with teeny tiny wav files, which I "amplified" so that I could work with them. I think it's because the mixer acted as a pre-amp, and made the signal too strong for the DP. Is that correct?

If the signal was 'too strong' you would not have a problem seeing them. The stronger the signal, the louder and bigger the waves are.

 

[SouthSIDE Glen]

lifelyrics,

When you get a few free minutes, grab yourself a nice cup of coffee or tea, and head over to www.independentrecording.net. Down in the lower right-hand corner of the front page is an icon labeled "Metering and Gain Structure'. Click on that to pop up an interactive graphical tutorial that covers the subjects of metering and signal level on one hand, and gain structure along the entire signal path on the other. It's a comprehensive tutorial with a lot of info, but it's all fairly basic, entry-level stuff only, assuming very little (if any) previous knowledge on the part of the reader.

G.

 

[lifelyrics]

You guys are awesome. Thanks so much.

Actually, after talking with my son (who helped us those couple of weeks we were trying to prepare the vocal tracks), he remembered a couple of things I forgot. (I'm definitely heading over to become more informed at the website you recommended, anyway)

We used the converter before we found the audacity "reverb" effect to be unacceptable (DP only has 1/4" inputs). After I sent her the tracks with said effect on them, we decided to completely re-record the vocals - it turned out better the second time around anyway. The second time around, we ran from Sennheiser dynamic mic -> Mackie mixer -> DP & keyboard amp (used as our monitor), simultaneously. When I mentioned the thought I had about the mixer acting as a pre-amp, Lukas remembered he had the gain set for what he would normally do in a live setting, for our church (more than half way). When we began recording, he was adjusting the input setting on the DP, because it was overloading so badly, before we even got to the chorus.

I have a lot to think about, and to learn. I am sure it won't take me long to research the necessary aspects of what I want to do, to make my future projects much more successful! I would be willing to send (via email) my recordings from the memorial service to any of you who are interested, and would like to offer any more feedback, based on what you hear.

Thanks again, guys.

 

[gecko zzed]

Two great terrors of recording are noise and distortion. These two assassins of music hide behind every door of the recording process, from source (your voice) to destination (the listener's ear).

Consider, for example, the space in which you record. If it is a well-designed, purpose-built space, there will be little (if no) ambient noise. However, many people record in bedrooms, lounge rooms, garages . . . wherever they can get away with it. Present in these spaces are an assortment of noises, e.g. fridges, computer fans, air-conditioners, the neighboor's rooster, cars going down the road. Additionally, there can be distortions, e.g. the room dimensions and construction could be such that some frequencies are unpleasantly accentuated.

If you can't eliminate these, then you have to manage them. Microphone technique is one way of doing this. You need to maximise the source (e.g. your voice) in relation to the other acoustical hazards. Two common methods are to sing louder or to get closer to the mike. However, these two can introduce their own problems, such as overloading the mike because of too strong a signal or introducing popping on the plosives (the 'p's and 't's, for example).

So the art here is to find the compromise . . . getting close enough and singing loudly enough to overwhelm the ambient noise without going too far and overloading the next link in the recording chain. This is where level meters and clipping indicators become your ally, with the idea of creating a strong signal without inducing clipping. Every device in the recording path shares the characteristics of your room, i.e. there is ambient noise (e.g. hiss and other electronic noises), and a point above which you introduce distortion. And this is where you can get help from Glen's website. In every step of the process there are ways of finding the path that threads its way between these two problem areas. I note, though, that not all devices are born equal. Some have lower noise floors than others, and some have more headroom.

However, I now add one more complicating factor. Getting close to the mike may reduce the impact of background noise, but could (even without distortion) produce an unsatisfying result. This is because not all things are amenable to close-miking. For example, a guitar is very big in comparison to a mike capsule, and radiates sound from all over its body. A full guitar sound can't be appreciated without the listener (or the mike) being some distance away from it. If you close-mike a guitar, you can miss this fullness because you are only focussing on a part of the guitar. One way to overcome this is to have two or more mikes on the guitar, each capturing a specific area. Or placing the mike some distance away . . . then suffering the consequences of the greater impact of ambient noise. Adding to this is that some instruments (e.g. a flute) sound best when the space in which they are played is allowed to form part of their sound. However the space needs to be very amenable to this, and in many domestic situations, it isn't. This usually means close-miking and adding the ambience (e.g. reverb) afterwards.

The compromise is now keeping the signal above the noise floor, not allowing it to distort AND preserving musicality.

 

[SouthSIDE Glen]

So the art here is to find the compromise

Great post, Mike!

Compromise is indeed what it is all about. Or, if I may put a slightly different spin on it, finding your preferred balance between cost and benefit. Cost in terms of quality-limiting factors such as noise, distortion and dynamic range, benefit in terms of achieving "that sound" that we desire. Quality audio engineering in large part involves learning how to "game the system" to get the maximum benefit for minimum cost.

G.

 

[lifelyrics]

I've printed this, and several other threads to my OneNote, so that I can easily refer to all of this very useful information. I'm going to take the next few days to absorb all of this, while experimenting with my new equipment.

My dad always used to tell me to understand my question before I asked him something, to discourage me from being spoon fed an answer for short term use. I'm at the point now, where I've asked several preliminary questions, and need to digest all the information so that I can understand the next set of questions I will surely be asking!

I am truly grateful for all of the help I've been offered at this website.