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Gear / Technical Help => Recording Gear => Topic started by: Gutbucket on February 20, 2008, 05:46:58 PM
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Two part discussion. Apologies for the lengthy prelude, but I want to try and frame it clearly-
I'd like to to discuss what is actually happening 'in the box' when choosing between the mic-input or line-input jacks on the portable recorders we are using around here. This thread is a result of some discussion in the Edirol R-44 recorder thread (http://taperssection.com/index.php/topic,97214.0.html). Specifically here (http://taperssection.com/index.php/topic,97214.msg1310236.html#msg1310236) and here (http://taperssection.com/index.php/topic,97214.msg1310546.html#msg1310546) and here (http://taperssection.com/index.php/topic,97214.msg1321643.html#msg1321643). If we can better understand how these boxes accommodate signals of varying level, we should be able make informed choices about what settings and even which inputs are most appropriate to use on recorders, external gear (such as preamps or in-line attenuating filters) and on microphones.
But that leads to the more basic discussion of how to optimize the signal path in order to make the most of the available headroom in each stage while minimizing noise and distortions. When running our gear, most of us find a sort of sweet spot in the configuration and settings that we are comfortable with through trial and error. After a while we learn the particulars of each piece of gear and can then get everything pretty close to spot-on just by knowing what the recording situation is, with some minor adjusting as things get rolling.
If you only have mics without switches on them and run them straight into the recorder then there aren't many variables to worry about other than making sure the input and gain settings you are using can comfortably handle the signal level from the microphones.
If you have microphones with pad switches running into an external preamp with adjustable gain settings feeding your recording device with multiple input choices, gain potentiometers and level switches (maybe some attenuators in your gear bag, too), the variables are near endless and the optimal settings are not as straight forward. You can adjust levels at the mic, at the external preamp, at the recorder, or in between in various ways.
So on a basic level, I'd like to discuss ways we can we can determine what should be the optimal gain and attenuation settings for the whole signal path, as opposed to just figuring it out empirically based on trial and error and then going on past experience (not that there's anything wrong with that). I don't mean to insult anyone's level of knowledge, here are a couple basic starting points of the empirical 'past experience method' - Generally, if using an external preamp, the preamp is set so that most of the necessary gain is made in that stage since usually the external preamp's quality is better than the built-in preamps of the recorder. Typically its a good idea to try and stay away from the extremes of maxing out any adjustment on one stage just to turn it down on another. Oversimplifying, maximum gain settings can often result in 'noise' and perhaps 'overload', minimum settings can also mean 'noise' and can cause the prior stage to work too hard and 'overload'. These things are not always 'in-your-face' distortions, but can be subtle low level degradations that can be easily overlooked (overheard? no that's the opposite). I guess what I'm getting at is that the optimal settings may not always be obvious even to the experienced.
The more specific question that spurred this thread is: What is actually taking place signal-wise in these small recorders? As I understand it, most if not all of these portable recorders (such as the Edirols) do not have separate line and mic preamp stages. Many have an analog front end and then a selectable digital gain stage (to choose between high or low sensitivity on the mic input for instance). Obviously the lower digital gain setting will be less noisy, but the choice between using the line-input or the low sensitivity mic-input is not as obvious without knowing if the recorder is either adding more gain for the mic input or just switching in a pad for the line-input. Does going line-in just add a pad to the mic-in and continue through the same preamp stage or does selecting the line-input actually reduce the gain added in the preamp stage?
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Great topic.
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One thing Doug Oade told me last year was that the new generation of recorders put out over the last 2-3 years lack mechanical input selection switching to distinct signal paths and have active electronics between the physical input connection, and the gain control.
This means that even line in, recorders can have input overloading even though the gain is set where the levels on the recorder are showing below 0db peaks. I know because I had a Panic recording look fine at the show, and end up clipping these circuits Doug mentioned.
This is not the case with the Apogee and Grace Pre/AD combo boxes, or the SBM-1 that I still use. If the levels show not going over 0db, there is no overloading happening on the older designs.
I think in the era of the ipod and cell phones of today, toggle wheels and other electronic menu style input selection seems better, but it's not.
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Not much active discussion on this tough topic.. here's some input from DSatz over in the Trim knobs and Using them (http://taperssection.com/index.php/topic,99365.0.html)thread that seems appropriate to quote here.
> ( ... my question applies to any device with trim)
There's no one general "right answer" because sometimes the thing which is supposedly just a "trim" control is actually the overall sensitivity control for the input stage; other times it really is just a small variable boost later on in the circuit, and is mainly there to help you optimize your recording levels, as you might expect.
You need to set good levels without overloading any stage of a recorder (input, intermediate, final, whatever), but the meters usually only tell you what's happening at the final stage of recording (i.e. at the input of the A/D converter). That's always a problem, and with different sets of equipment it has to be worked out differently. Sometimes you have to pad a signal before sending it into the recorder in the first place (like when people get clipping even when their peaks are only at -10 dB).
--best regards
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+t for opening a fascinating can of worms.
I've nothing to contribute here but will be reading with much interest...
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Helo,
When I first saw this topic I thought to myself: that's the one, that's the kind of discussion I was waiting for. But I need more! ;D
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I am interested to learn about running the pad on the mics vs on the recorder.
Reading the information about electronics overloading before they even get to the meters makes me wonder if running the pad on the mics is a good idea....
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I am interested to learn about running the pad on the mics vs on the recorder.
Reading the information about electronics overloading before they even get to the meters makes me wonder if running the pad on the mics is a good idea....
I'm very interested in this as well. Maybe this has been covered elsewhere, but I'm specifically curious about others' experience using the built in pad on 480s vs. other methods for attenuation. To date I've only recorded one show where running the mics into my R-4 (Oade's ACM on channels 3 & 4) resulted in clipping at minimum gain. I decided (after soliciting a more experienced taper's advice) to switch in the pad on the mics instead of using the R-4's "line-in" -20dB attenuation or the XLR attenuators I have on hand for taming hot board feeds, figuring it would be the optimal way to bring down the signal. But that pull really doesn't sound as natural to me as my other 481 DIN recordings -- hard to describe, but it has kind of a slightly "underwater" feel to it, as if some kind of veil had been stuck between the music and the mics. This could possibly be a byproduct of the high SPLs slamming the CK61 diaphragms however, as I don't usually record shows at that volume level and really have no basis of comparison.
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I'll try and get some of my thoughts together in a post over the next few days when I get a chance. I don't have answers, but I have somewhat of a grasp of the issues.. and loads of questions.
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I'll try and get some of my thoughts together in a post over the next few days when I get a chance. I don't have answers, but I have somewhat of a grasp of the issues.. and loads of questions.
Loads of questions? Same thing here. ;)
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My deck goes to 11 and my tapes are heaven!
Talk is cheap. I think the key is rigorous testing to find what sounds best. For example, we have yet to see a comp that compares running the gain of a 7xx with a pre in front at 0db vs. -6 db. And many of us run that combo so we have a lot invested in the outcome.. I once read where Moke said he runs at -6 with the v3 in front, I run 0. We've had debates over optimal peak and rms levels on the 7xx. I don't think that has been settled.
We can talk about the 722's input stage, impedance, reverse engineer, etc, all day long but only a test and comp is going to get to the all important 'what sounds best'.
In the case of the r09, more than a few of us spent time trying to measure unity and stuff like that. What we missed for quite a while was the bass distortion when run with the gain down around 1-6. That distortion occurs with levels peaking at only -6dB, so levels on the recorder don't even have to be close to 0. So that's a case where just trying your gear at different gain settings with pre-recorded bass-heavy material would have revealed pretty serious matching flaws at certain settings.
I agree on the mic pad question... My psp2 has a fixed gain of about 25dB. With the MGs I run the mic pad for loud shows. Can't help but wonder if other attenuation would sound better. I can't run the schoeps in those situations due to a lack of a built in pad. My calculations last year for Dinosaur Jr. showed that the levels were beyond the SPL handling of the schoeps. So even with a pre-amp that can be run to add no gain (like the aerco or v3 with the -20 pads engaged), you're 'out of the envelope' at the mics...
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I am interested to learn about running the pad on the mics vs on the recorder.
Reading the information about electronics overloading before they even get to the meters makes me wonder if running the pad on the mics is a good idea....
I'm very interested in this as well. Maybe this has been covered elsewhere, but I'm specifically curious about others' experience using the built in pad on 480s vs. other methods for attenuation. To date I've only recorded one show where running the mics into my R-4 (Oade's ACM on channels 3 & 4) resulted in clipping at minimum gain. I decided (after soliciting a more experienced taper's advice) to switch in the pad on the mics instead of using the R-4's "line-in" -20dB attenuation or the XLR attenuators I have on hand for taming hot board feeds, figuring it would be the optimal way to bring down the signal. But that pull really doesn't sound as natural to me as my other 481 DIN recordings -- hard to describe, but it has kind of a slightly "underwater" feel to it, as if some kind of veil had been stuck between the music and the mics. This could possibly be a byproduct of the high SPLs slamming the CK61 diaphragms however, as I don't usually record shows at that volume level and really have no basis of comparison.
I started running my 460s at "0" in '94, and in the summer of '95, I was talking about the AKGs to Doug at a show somewhere and he explained that AKGs are unique with a sensitivity switch, rather than a traditional attenuation resistor pad. Setting them to "-10" does not introduce any resistance, it simply drops the sensitivity by -10 db at the preamp stage. This allows the 460s (and the current 480s) to take 144db sound pressure with no other changes in signal quality. I made the switch and never touched them again, getting the maximum headroom possible from the mics. For PA taping, this really made a difference since AKGs are already hot output mics to begin with.
The headroom of the mics is to me the first and most important place to start, since distortion at any point in the signal chain continues to the end result.
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Good point, Freelunch. Talk is cheap, experience isn't.
And actually, that makes me realize this is really a three part discussion. First is the broad general principles involved, second is the applied engineering (understanding how to do the math, converting & matching specs, etc), third and most specific is the on-hand experience with particular gear like you mention above. You can jump straight to the third part and just use your ears and probably develop a feel for the general principles (if you didn't have much of one to start) because you of course need to use your ears as the final judge in any case, but it helps to know what's going on. I personally need most help understanding the mathematics part. When I get to that point in a post or two you can hopefully school me big time. Any wise fool with good ears can skip the analytical engineering method and figure out the best sounding combination of settings given enough time and trial. That's what I do too, but I personally would like a deeper understanding.
The general principles could & should be much better stated by someone else, but since I opened the can, here goes this layman's version:
Reminders: We're talking only about signal level management here. I don't mean to insult anyone's intelligence, but I do want to lay down some groundwork that may be blatantly obvious to some, for the benefit of the less experienced. just skip this post if you know all this..
Nothing in this mortal world is of infinite bandwidth. Not the air carrying the sounds themselves, not our ears and not any electronic gear capable of being built. All systems have limits at both ends of the spectrum of quiet to loud. Narrowing down that big picture to the equipment we're using around here, each and every piece of gear that the signal passes though has a limited range of signal levels where it performs well. That includes cables too, but let's ignore them since by this measure, cables are going to outperform every other piece of gear by a very large margin. In fact, each piece has two limited ranges- a range of input signal strength and a range of output signal strength. Get t0o close to the bottom of that range and things dissolve into noise, get too close to the top of the range and things dissolve into noise of a different type. Hopefully the full range of the signals we want to record will fit within the range that our equipment can handle comfortably. What we need to do is be aware of the limited range of each stage of our recording chain and adjust things to fit without getting too close to either extreme. We need to do this for each component in the chain.
Start with the mics. A mic's input range restriction is the range of sound pressure (SPL) that the mic can handle. It is grossly determined by the self noise specification at the quiet end of the scale, and the maximum SPL level rating before clipping at the loud end of its range. Being mechanical and electrical devices, mics are actually several systems together in one package. First there is the mechanical system of the capsule housing, the movement of air in it and the vibrating diaphragm. The diaphragm's signal limits are determined by the physical constraints of its design. At the lower end the signal will disappear into the noise of random air molecules impinging on it's surface, at the upper end the diaphragm will reach the physical limit of it's travel. There is nothing we as end users can do to change this range except to choose a different microphone or move to a different location with a more favorable range of loudness.
The movement of the diaphragm is converted to a weak analog electrical signal (how that's done varies by microphone type and isn't important here). Condenser mics are not capable of driving the next piece of equipment in the chain without that signal being converted to a more suitable one by the electronics in the mic capsule, mic body, or both. The mic's electronic stage can be simple or more complex, but either way that stage has it's own signal range, separate from that of the diaphragm. Some mics allow for adjustments to that range, and effectively give up some of one end of the range to gain some room at the other. This can be done in different ways. It may be done by switching in a 'pad' or resistance that lowers the level going though this stage, giving up some of the quiet end of the range to gain loudness room up top, or as DATBRAD mentions, the level of amplification in the mic body might be changed to effectively do the same thing (with the same basic trade-off in range, though possibly audible changes like Fred reports from his experience with the AKGs.. but that's for part 3). I don't know if this pad is typically inserted before or after the electronic stage or if the electronic stage usually has as great a range as the mechanical diaphragm does. Why does that matter? Because if the input range of the electronic stage is less than the output range of the diaphragm, engaging the pad on the mic may keep the electronics in the microphone from distorting at loud levels (I suspect this is the case). However, if the input range of the electronics is greater than that of the diaphragm there is no worry of clipping the electronics first and the pad would just lower the output level of the electronic stage to the next component. In that case the mic would only distort once the SPL reached the input limit of the mic regardless of the pad setting and the pad would act exactly like an in-line attenuator of equivalent reduction value placed between the mic and mic cable. After the mic electronics stage the signal is ready to be passed to the next component in the recording chain.
At this point the signal level has been conditioned and boosted enough so that it can be transferred to the next component in the system by a mic cable. The reason that 'active' cables are so expensive is that they are transferring the weak, 'not suitable for travel' signal directly from the capsule to the electronics in the mic body. Before that signal can be recorded it probably needs to be adjusted yet again by a preamp stage that is either built into the recorder or an external unit (sometimes both). If using an external preamp, that unit has a limited range too. If the signal going though it is too strong it may clip and distort. If too low it may need to be cranked up to add lots of gain - maybe more than the preamp can comfortably provide without adding noise because the signal was too close to the lower limit of its range; maybe it doesn't have the capability of adding enough gain; or maybe it's gain adjustment is limited or not adjustable at all. To complicate things, the external preamp is often also doing other jobs like powering the mics so it may not be an optional component even if the gain it adds is unwanted. In any case, the preamp will be most comfortable when run with a certain input range from the mic and a certain output range to the recorder. If the signal out of the microphones is too hot for the preamp's input range you could:
1) engage the 'pad' on the mic to lower the signal if there is one.
2) add an in-line attenuator between the mic and the preamp.
3) engage the 'pad' on the preamp if there is one.
What are the tradeoffs? Well if you have a choice and if the mic's output is within its comfort range, it would probably be best to attenuate after the mic and after the mic cable. Doing so would allow a stronger signal level through the mic cable which means any electronic interference noise picked up in the cable itself will be proportionately lower in level than the signal you're tying to capture. You could do so by using the input pad on the preamp if there is one, or by using an in-line attenuator between the mic cable and and recorder. I can't see any reason to put an in-line attenuator before the mic cable, unless its for reasons other than optimizing the signal path (like if the recorder won't fit into your bag with the attenuators sticking out the side :P).
That's enough for for now, more on the preamp stage next time.
Mic gurus, EE types and those with more experience equity than me, feel free to correct anything so far.
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Great thread.
I started running my D5 again and have been wondering about signal management and maximizing headroom at the mic-pre. When going mic-in I always use the 20db attenuator when recording PA systems. I used to have a pair of 10db in-line attenuators (ProCo) because the 20db pad on the deck was not enough (still had just a little bit of overloading) . So, would running the above setup be the best route or would it make more sense to use a 30db attenuator and skip the pad on the deck? I am using the Sennheiser ME-40 mics (no pad).
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Ugh, this thread lives. Honestly I don't know.
I felt that in trying to really boil it down I just made a big blathering mess, bored myself and others reading it and got over my head trying to simplify complex interactions. :-\ I was working towards doing the calculations to analyze the headroom in the various stages of my equipment chain, but I gave it up for now. I can set everything up by ear without working so hard. I still wonder about what's going on out of philosophical curiosity, but I feel like I got all bogged in lecture-y toned posts above in what was supposed to be an attempt at laying a philosophical groundwork.
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Quoted from another thread, but exactly what I was working towards here-
... When sound is so loud that it threatens to overload the microphone's own electronics (a capsule is nearly impossible to overload with sound alone), you have to use the pad in the microphone. In effect, it operates between the capsule and the "amplifier" of the microphone.
But the pad in a microphone, with very few exceptions, raises that microphone's noise level relative to its sensitivity, since it reduces the output of the capsule at all levels without affecting the (fixed) noise level of the mike's electronics. The pad at the input of a recorder generally doesn't have this problem, so if you can use the pad in the recorder, it's generally preferable.
If a recorder is being overloaded with signal from the microphones, and the recorder has no pad of its own, then it's generally better to buy in-line resistive pads and use them at the inputs of the recorder, rather than use the microphones' pads for this purpose. In a pinch, however, a moderate amount of extra noise is preferable to overload distortion ("brickwalling")--so if you have absolutely no alternative, use the pads in the microphones.
Most microphones nowadays can handle such high sound pressure levels that the only times they overload are when the sound levels are actually hazardous to your hearing--or else something else is going on, such as wind noise or inadequate powering.
--best regards
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Thanks for that, that's good info...
Now, what kind of started some of this was my use of my mic's -10db pad in order to avoid using my R4's -20db pad. The trade-off sounds like it's an "add more noise by using the mic's pad and less noise from the R4's preamps b/c you don't have to use as much gain" vs. the "don't add noise by using the mic's pad but b/c the R4's pad is bigger at -20db you add more noise by having to using more gain on the R4 preamps." I'm also a little curious about whether using a larger pad (in general and all other things being equal) is worse than using a smaller pad.
But beyond that, and upon reflecting more on DSatz's comments, it seems like I might want to reconsider using the R4's pad a little more often.
One other question I have -- given what DSatz just said -- is that as an owner/user of AKG480's, I actually have a +6db setting on the same switch as the -10db pad (it's +6, 0, -10). I wonder if by using that, I can actually raise my signal higher off the noise floor, which would be great. If that's true, then it'd be worth doing as long as the signal doesn't overload the mic's built-in preamp in the body. Considering I record a lot of acoustic jazz, classical, and chamber music, I might be better off going +6 on the mic, and then go line-in (pad on) on the R4 if it's too loud for mic in. That'd get my signal higher off the noise floor by using the +6, then the pad on the R4 would keep me from clipping the recorder, and because of the +6 on the mics, I'd need less gain on the R4. This might be moot though, because I can usually run mic-in for a lot of my gigs (at 0 on the mics).
Interesting...
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With all the components, variables, and options, it's sort of amazing that ANY
recording turns out OK! :)
scot
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Attenuating later in the signal chain should also reduce low level EMI interferences picked up in the cabling and any other intervening equipment (such as a mic powering box). In general I'd think that the better option unless there is reason to fear of eating up all the desired headroom in some part of the chain before the pad. The other issue that is a mysterious, hidden bugaboo is "what is actually going on with the gain structure of these all-in-one recorders like the R-4?" Does changing input sensitivities on the box (via switch or input jack choice) mean switching simple resistive attenuation in and out, increasing or reducing an analog amplification stage, adding or removing an additional analog gain stage, or doing a digital signal adjustment?
I guess I haven't payed enough attention to the manufacturer's spec's of mics that have switchable pads either. Do their maximum SPL before clipping measurements indicate if those measurements were made with the internal pad engaged at it's most attenuating setting or flat?
It would be really interesting to setup a 'research' recording rig that included metering marked with the appropriate range between noise floor and clipping (or some other acceptable distortion level) at every stage of the chain to really get a handle on what's going on.
I imagine I could learn a lot experimenting with various combinations of settings and also get a great 'seat of the pants' feel for the level magnitude differences of the various stages.
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Pretty sure in the R4 that the mic/line switch is simply a 20 db pad, but YMMV with other devices.
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Ahh, I've read this thread a few times but just can't seem to nail down the answer I am looking for. When running Sennheiser ME-40s > Sony TC-D5 PROII I need 30db of reduction to avoid overload. I ca get away with 20db in quieter settings but FOB and rock'n'roll PA's necessitate 30db. So what is the best option? Use the 20db pad on the deck and a 10db inline pad or just use a 30db inline pad. I just want to get as much out of the D5's mic-pre as possible, and I don't know much about the quality of the 20db attenuator circuit in the D5 Thanks!! :coolguy:
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In that case, and from some of what I took from this thread, you might be better off with the 30db pad. The logic was that when using the pad on the mic, you are basically giving up some of your signal to noise ratio on the mic. The pad on the mic is really designed to help prevent overload of the mic body itself, it lowers the signal before its own circuitry. (I think, LOL.)
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In that case, and from some of what I took from this thread, you might be better off with the 30db pad. The logic was that when using the pad on the mic, you are basically giving up some of your signal to noise ratio on the mic. The pad on the mic is really designed to help prevent overload of the mic body itself, it lowers the signal before its own circuitry. (I think, LOL.)
Thanks for the quick response! :coolguy:
I have been following the idea behind giving up some s/n when using the pad on the mic - great info, BTW - but these mics have no onboard pad, leading me to the use of xlr in-line attenuators. So I am still not sure which approach is the best. 30db Attenuator and no attenuation on the deck, or 10db attenuator and 20db attenuator on the deck? :hmmm:
I may be over thinking all of this- just want to maximize s/n and get the most out of this little beast!
Thanks again