dex Otaku

Hints: How to Control Your Levels and Make Undistorted Recordings

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Preface: As with most of the things I write, the length of this is sure to seem daunting. To those of you who are intimidated by such a thing, there's an easy solution: don't read it, and likewise don't bother to learn how to do a better job of recording. ;)

Please forgive the odd formatting [strange varying font sizes for no reason] – it appears to be something that the board's skin imposes on html blockquote sections, which I can't get rid of without manually setting the font size of every section. Bad CSS, BAD! ;)

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Member Navaro sent me this message last night [reposted here with permission]:

I've been a member of this board for a few months now...my main concern is---> you guessed it, distortion. like yourself, I was wondering what all the deal was with battery modules and attenuators.

. . .

with my MZ-RH910 I always record with the mic AGC setting to 'for loud music', SP-BMC-12 to mic-in, and Hi-SP. When I started using manual recording levels, without the aid of the attenuator, I'd set the levels to about 18/30. This was a bit too high as I've learned the hard way, I trimmed it down to 15, then 14 but the vocals would still distort.

. . .

one technique I have yet to try is going from mic to line-in instead of mic-in. one more thing I haven't tried yet is using the mic-->attenuator-->mic-in, using manual levels set to anywhere between 20/30 to 25/30. when I recorded my yeah yeah yeahs concert in april, I used mic-->attenuator-->mic-in with levels set at 15/30 (at this time I didn't realize that with the volume turned up all the way, it still was attenuated a great deal and so my recording came out very quiet. had I known this, I would have boosted it to 25/30).

There are a few questions I can round down from his messages and attempt to answer here. Anyone who can come up with more correct or clearer answers, please do add to this thread.

<ul><li>What is the difference between mic-in and line-in?</li><li>What is AGC?</li><li>What is the gain setting for and what does it affect?</li><li>Why should I use manual levels instead of AGC?</li><li>What level should I use when recording with manual levels?</li><li>What is a battery module for?</li><li>What's the practical difference between a battery module and an attenuator?</li><li>What's an external preamplifier for?</li></ul>

Some of my responses contain the answers to the other questions, so there is repetition in here for those who actually read the whole thing.

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<blockquote><i>For those who are interested in more details about the quality of HiMD recorders' mic preamps, see this article at the Wildlife Sound Recording Society's website, and its two related articles, noise performance and gain and overload performance, as tested with an NH-700.</i></blockquote>

<hr noshade size=1><blockquote><b>[note: all of the following applies only to recording via analogue inputs]</b></blockquote>

<b>What is the difference between mic-in and line-in?</b> <blockquote>The first difference between these two is that the mic input has a preamplifier made to bring microphone-level signals [millivolts-range] to line-level [around 1V peak-to-peak]. Most MD and HiMD recorders have a low and high-gain [or sensitivity] setting. Under most circumstances – unless the sounds you're trying to record are rather quiet [since high-sens adds another 20dB of gain to the preamp] – the low-gain setting should suffice.

The second difference between the two analogue inputs is that the mic-in has “plug-in power” on it. Plug-in power is meant to power electret condenser microphones such as the SP-BMC-12s being used by Navaro [or the similar SP-TFB-2s I use]. condenser microphones require bias power in order to work; HiMD recorders supply between 1 and 2.5V on the mic input for this purpose.

The line input is made to take line-level signals [around 1V peak-to-peak]. It has a very limited amount of gain as it is made to take signals that have already been preamplified for use with a recorder or amplifier. Because there is no “plug-in power” on the line input, electret condenser microphones will not provide any signal if plugged in there.</blockquote>

<b>What is AGC?</b><blockquote>AGC means Auto Gain Control. Some companies refer to the same thing as Auto Level Control [ALC] among other things.

The basic idea behind AGC is that when the volume get too high, the recorder does the job of “ducking” levels automatically to avoid distortion in your recording.

In more technical terms, AGC is a basic implementation of a compressor/limiter, or [more likely, in my opinion] a dynamic compressor [that being a compressor whose ratio increases as levels increase above its threshold].

There are two modes with Sony's AGC on MD and HiMD recorders: Standard and LoudMusic [or ForLoudMusic depending on your model]. The most likely difference between these two settings [i could be wrong on this, but this is my guess based on the difference in how they sound] is respectively a short release time versus a longer release time. In situations where the volume is consistently loud, the difference between them shouldn't be very obvious, but in situations where the volume varies greatly over short periods, LoudMusic is likely a better choice since the compression taking place should be less obvious [by avoiding the “pumping” effect of a fast release].

<b>The advantage with AGC is that it's a relatively “fire and forget” way of doing things. You can start recording and never mind setting levels or monitoring the meters.</b>

<b>The disadvantages to AGC mostly revolve around having to rely on how the fixed gain of the mic preamp relates to whatever the sensitivity of your microphone is, as well as the difference between the softest and loudest sounds [dynamic range] you're recording.</b>

That may sound complicated, but what it amounts to is this: <b>using AGC means you're stuck with a default that doesn't always work, and in many cases will make things substantially worse rather than a bit better.</b> This is another prime example, in my mind, of something that makes things much harder in the end even though it's intended purpose it to make things simpler to begin with.

<b>In some cases, the default works quite well. For example, </b>with my SP-TFB-2s, with the mic preamp set to low-sens, the average level of conversation falls just below the AGC's threshold; this means that any sounds ranging from quiet to conversation-level are uncompressed and sound completely natural, whereas anything much louder than conversation gets compressed/limited to prevent distortion. This is pretty much an ideal case for using the AGC.

<b>Another case in which the AGC works well is for what I would call broadcast-ready recording</b> [hopefully no broadcast recordists take offense to this]. In these situations it's usually more important to capture the sound, period, than to capture the sound with high fidelity and a natural dynamic range. Using a monaural omnidirectional microphone and the high-sens setting, a reporter's recording captures all of the sound in a media scrum or of an interview. In this case, though – all of the sound is highly compressed, and it's obvious when it's played back. Again, this is pretty much an ideal case.

<b>Most live+amplified music recording situations work <i>very</i> poorly with AGC. </b> Acoustic recordings can work great if you're distant-mic'ing, but highly-amplified music or even mildly-amplified in a small space can quickly push levels into solid compression, basically eliminating <i>any</i> sense of dynamic range in the recording. The results tend to sound quite unnatural, though of course this is as much a matter of preference as anything; some people actually like their recordings to be constantly and consistently at top volume. For some listening situations, this is actually appropriate, too – such as when listening in a noisy vehicle, or with very cheap, low-power portable stereo systems.

As a last note to this section – be sure to read the section <i>When recording with the line-in:</i> at the end of my answer to <i>Why should I use manual levels instead of AGC.</i></blockquote>

<b>What is the gain [mic sensitivity] setting for and what does it affect?</b><blockquote>The gain setting sets the mic preamplifier's gain. High-sens mode adds another 20dB of gain compared to low-sens mode.

With most of the microphones we use, the following tend to be true:<ul><li>The low-sens mode is appropriate for most recording conditions with sounds ranging in level from conversation to loud music</li><li>The high-sens mode is appropriate for recording quiet sounds or as suggested in the example above [broadcast-ready recording]</li></ul>

There is considerable overlap between the two ranges if you're using manual record levels, but with practise it becomes reasonably easy to recognise which is better for a given situation with your equipment.</blockquote>

<b>Why should I use manual levels instead of AGC?</b>

<blockquote><i>When recording with a microphone:</i><blockquote>First off: this is as much a matter of personal opinion as it is something that depends on exactly the conditions under which you're recording.

<b>The biggest reason I can give is that judicious use of manual levels will <i>always</i> give a more natural-sounding result.</b> This is a more purist [dare I use that word] approach to recording since the first generation is thus completely unprocessed [which is always a good thing, in my opinion].

<blockquote><i>Side-note:<a href="http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=59">How To Make Better Recordings in the 21st Century--An Integrated Approach to Metering, Monitoring, and Leveling Practices</a>, from the <a href="http://www.digido.com">Digital Domain website.</a></i></blockquote>

The next biggest reason is that we're recording with digital equipment, here. <b>Given that the 16-bit quantisation gives a usable dynamic range of 96dB, and that most of the microphones that HiMD recordists tend to use have self-noise which is louder than the mic preamps in their recorders [the total usable dynamic range with HiMD is above 85dB], even if you crank your recording levels <i>down</i> you're still going to end up with a recording that has a low noise floor and a natural dynamic range.</b>

A prime example here: professional digital recording equipment's VU meters follow one of several reference standards for measurement. My preferred measurement standard is that established by the European Broadcast Union [EBU – which I've seen used by most film recordists, who follow the most rigid standards in the world of audio], which sets 0VU at around -20dBfs. Yes, you read that right .. <b> -20dBfs</b>.

Going by the good old “standard” methods of record-metering, that means that you ideally want your average level while recording to be at 0VU, and the case of EBU metering, that gives you a whopping 20dB of headroom.

Even the record meters on MD and HiMD are relatively conservative in this regard: that centre hash [consider it the 0VU mark] on the meter is -12dBfs. Directly from the RH10 operations manual:

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What does this really mean to recordists?

It means – if you set levels manually according to your recorder's meter, with an average around the 0VU mark, you'll have at least 12dB of headroom. That means 12dB more volume above what you've measured for <b>without distortion</b> [taking into account the limits of both your mic and preamp, that is].

I, personally, try to average a bar or two below the HiMD 0VU mark, giving about 20dB of headroom.

Of course, when you get this home, upload your recording, and play it back – it's going to be pretty quiet compared to your favourite recent pop recordings off CD. Scroll back up and read the article at Digital Domain. You'll see why – it's because your monitoring [i.e. your amp+speaker setup] is set to listen to “average” listening material at “average” volume, which [in my opinion, though it does depend on what the content is] doesn't reflect good recording [or mixing, or mastering] practises in any way, shape, or form.

So – to try and sum this up, a suggestion to recordists: <b>you have a digital recording medium, with a decent mic preamp and a huge dynamic range – <i>****USE IT!****</i> Save the bit-pushing for the mastering stage, if you must do it at all. After all – GIGO – Garbage In, Garbage Out. If your recording starts off compressed or distorted, then it will end up compressed or distorted. </b>

<b>The lesson is – USE YOUR RECORD METERS. If you have a non-backlit LCD display, carry a LED flashlight. Or, if you have to pocket your recorder before heading into a concert because you're “stealthing” </b>[i'll leave the lecture on honouring IP/copyright to someone else this time], <b>set your levels conservatively</b> [noting that knowing what level is “right” for your equipment is something you have to learn by experience], <b>and learn to use the tools you have in post-production</b> [such as non-linear audio editing software with settings or plugins for gain and dynamics processing].

<b>Likewise, there are cases when you <i>know</i> very well that what you're recording is simply not going to get much or even any louder than what you're metering to begin with, so there's less need to leave massive headroom.</b> Use your own best judgement, and remember another old saying – <i>better safe than sorry.</i>

If you go for the cleanest possible recording to begin with, you'll have a first generation that is highly usable in the editing and mastering stages, that you can process any way you like – including adding compression, EQing that bass that was too loud because of your cheap microphones, and what have you.

As a last note – using manual levels is not perfect for every situation. There will be times when you'll need way more headroom than expected because certain sounds will be very loud [resulting in clipping distortion if you don't or can't address the problem]; there are others when your average level is quieter than expected, and you'll find the [albeit usually rather low] noise floor audible after editing.

The solution here is to experiment and figure out for yourself when it's most appropriate to use one method or the other – not to mention how with practise, you'll learn where to set levels with your equipment setup simply “by ear.”</blockquote>

<i>When recording with the line-in:</i><blockquote>The same basic principles apply with line-in recording as for mic recording, but there are a few reasons why sometimes it's appropriate, not to mention more convenient, to NOT use manual level control.

A perfect example of this: while at a show and making a recording directly “off the board,” I almost <i>never</i> use manual levels. Most mixing desks either allow you to plug into your recorder through a separate “tape out” feed or by using one stereo bus just for recording. Either way usually allows you to set the output level before it hits the recorder – meaning you can set your recorder running without ever touching its level controls, and then set the output level of the board to read with a good average – while still leaving plenty of headroom – according to your recorder's meters.

This has a pretty serious advantage to it: if you leave plenty of headroom, you'll still capture the dynamic range of the performance, but – you'll also have the AGC there to catch things should levels suddenly jump too high. This is the best of both worlds, really: you've set your levels manually [using the controls on the sound board], but you still have the safety-net of the AGC there to prevent outright clipping distortion should things suddenly jump in level.

Ideally, Sony would enable this as an option on their recorder – after all, the hardware is already in there to do it – that is, to use manual levels but still have the bonus of overload protection. A menu item for “manual levels limiter” or something sure would be nice, especially considering the fact that this is the main reason so many inexperienced recordists try manual levels and then decide to do without – because as it is now, it's totally unforgiving [i.e. it results in unrepairable clipping distortion] if you don't leave sufficient headroom.</blockquote></blockquote>

<b>What level should I use when recording with manual levels?</b><blockquote><b>There is no straightforward answer to this question.</b> Every microphone [model] has a different sensitivity, every venue or location has its own acoustical characteristics, and every amplification system has its own volume. We can't guess as to how loud the PA will be at your local hole-in-the-wall punk venue, nor how loud an acoustic act will be from the fourth-row centre table at your local jazz club.

Not to repeat myself, but <b>USE YOUR RECORD METERS. If you have a non-backlit LCD display, carry a LED flashlight. Or, if you have to pocket your recorder before heading into a concert because you're “stealthing,” set your levels conservatively and learn to use the tools you have in post-production.</b>

When you drive, you don't just steer blindly; you [hopefully] look where you're going. You also look at instruments like your speedometer, and check your mirrors to see what other traffic is doing. To think that doing the best possible job of recording should be fire-and-forget is, I would say, unreasonable.

<b>It takes some effort to do a good job, but that doesn't mean it has to be difficult.</b></blockquote>

<b>What is a battery module for?</b><blockquote>Battery modules are for powering <a href="http://en.wikipedia.org/wiki/Microphone#Capacitor_or_condenser_microphones">condenser</a> and, more specifically in most of our cases here, <a href="http://en.wikipedia.org/wiki/Microphone#Electret_capacitor_microphones">electret condenser</a> microphones. Most other types of microphone do not require power in order to function, and some contain their own power supplies [such as the single AA battery in used by the Sony ECM MS-907, or the 9V battery optionally used by the Rode NT4 when you can't supply it with 48V phantom].

Different microphones have different bias voltage requirements. Stage and studio microphones tend to use 48V <a href="http://en.wikipedia.org/wiki/Phantom_power">phantom power</a>; location-recording equipment used by film sound recordists and broadcasters often use 12V phantom power. These are the two most common formats used by professional [balanced] equipment.

Smaller, portable microphones as usually used with MD and HiMD recorders work in the same fashion, but due to the requirement of portability, are made to work optimally with lower bias voltages. Most of the elements we use are made for an optimal bias of around 10V, ideal for use with a 9V battery. The 1-2.5V supplied from most HiMD recorders is sufficient to make them work, but isn't quite enough to expect full performance from the capsule. What is full performance, then?

The result of underbiasing a condenser mic is that the maximum SPL [loudest sound] it can transduce without distortion is reduced. A microphone that claims to have a max. SPL of 120dB [at its rated 10V bias] can fall to 105dB when biased at only 1.5V, for example.

In short, what a battery box does is make up this difference. In the case of the above microphone, its usable maximum SPL is increased by around 15dB simply by supplying the higher bias voltage.

For those who are recording loud rock concerts, this means the difference between registering a clear, clean recording, and having a disc's worth of unlistenable, distorted garbage.

<b>Battery boxes </b>[battboxes] <b>do not address the problem of MD and HiMD recorders' limited mic preamp headroom which leads to [preamp] clipping distortion when faced with high-level signals directly from a microphone.</b> In these cases, either attenuation is required between the mic and the preamp [not using a battbox], or between the battbox and the preamp [using a battbox], or you have to go directly into the line-in from the battbox [assuming the level from the mic is high enough above the noisefloor].

See Reactive's thoughts on the topic of attenuators here at <a href=”http://forums.minidisc.org/index.php?showtopic=9069”>this thread</a>.

Also see <a href="http://forums.minidisc.org/index.php?showtopic=11254">Greenmachine's DIY instructions for a simple battbox</a> [scroll down past the DIY microphone post].</blockquote>

<b>What's the practical difference between a battery module and an attenuator?</b><blockquote>First, realise that what an <a href="http://en.wikipedia.org/wiki/Attenuator">attenuator</a> does is <i>throw signal away</i>; it causes deliberate signal loss. Generally speaking, this is something engineers and purists will frown on doing unless it's absolutely necessary.

The most often seen reason for using one is in the case of trying to capture a very loud rock concert with a microphone who maximum SPL is not being exceeded, but whose sensitivity is high enough that its output level overloads the preamp it's being plugged into. In this case, the attenuator reduces the output level of the microphone [it throws away signal] sufficiently that it no longer overloads the preamp, making a clean recording [of the loud parts] possible.

I'm going to repeat myself somewhat here for those who skipped the last section: battery boxes [battboxes] do not address the problem of MD and HiMD recorders' limited mic preamp headroom which leads to [preamp] clipping distortion when faced with high-level signals directly from a microphone. In these cases, either attenuation is required between the mic and the preamp [not using a battbox], or between the battbox and the preamp [using a battbox], or you have to go directly into the line-in from the battbox [assuming the level from the mic is high enough above the noisefloor]. [see links at the end of the above section]

Many users of this board use variable attenuators when recording loud concerts and report good results. I, myself, have never tried it and have also never had reason to, as the loudest sounds I've recorded [such as jets taxiing, artillery fire, helicopters passing directly overhead – all of those at an air show, I'll point out.. fireworks, and thunder from close-striking lightning] exceeded the max SPL of my microphones [powered by my recorder, not a battbox] in addition to clipping at the preamp. In other words, I can not speak from experience on this matter.

I would like to point out one particular concern of mine, however:

Putting an attenuator in line with your microphone causes a loss in the signal from the mic, and also should cause a corresponding loss in the bias voltage being supplied by the recorder. Since recording high SPLs cleanly is the entire point behind using the attenuator, this might with some equipment end up being an exercise in self-defeat; you're lowering the output of the mics so it doesn't overload the preamp, but you're also lowering the maximum SPL the mics can transduce without distortion.

Whether this happens or not depends on exactly the mic, its sensitivity at the supplied bias voltage, &c. - and, judging by the experience of forum members such as A440, it doesn't happen with the most commonly-seen equipment.

Point being: it's unlikely to happen judging by the experience of our forum's users, but it's not impossible, which is why I put it out there.</blockquote>

<b>What's an external preamplifier for?</b><blockquote>An external preamp allows the recordist to completely avoid the pitfalls of microphone underbias, a possibly-noisy built-in preamp in their recorder, and the possibly-limited headroom of the same built-in preamp.

Ideally, this is the best of all worlds. With the better models available [which feature controllable variable-gain inputs], you plug the external preamp's output into the line-in of your recorder, set it to unity gain [18/30 on HiMD recorders] or even just leave the unit in AGC mode, plug the mic into the preamp, and set the gain of the preamp [with a nice tactile turn-it-by-hand knob] so that the recorder's meters are happy.

The advantages of using an external preamp with a variable-gain input can be summed up thus:<ul><li>The probable lower noisefloor of the external preamp makes for cleaner recordings of quiet sounds</li><li>The higher bias voltage it supplies to your microphone ensures that high SPLs can be recorded cleanly</li><li>The higher preamp headroom also ensures that loud sounds can be recorded cleanly without having to incur signal loss before the signal is even preamplified, let alone recorded [i.e. it avoids having to use an attenuator]</li></ul>

The chief disadvantage to using an external preamp is that it means having another box to both carry and power, as well as more cables snaking about your body.

All preamps are <i>not</i> created equal, and some designs actually just make things harder and worse to control [read: any model that uses fixed-gain, DIP switches to set the gain, and lacks any kind of feedback regarding its output levels] rather than either easier or better.

This is another matter of personal preference as well depending on what kind of recording you do – a fixed-gain preamp might be perfect for certain situations, while being generally less versatile.

I would personally recommend using an external preamp for things such as nature, ambient, environmental, or acoustic music recording – generally all things where detail and dynamic range are of importance from the get-go. The potential lower noisefloor, probable higher gain and headroom of the preamp will make the investment worth it; also, since many of these types of recordings are made rather deliberately, the inconvenience of the extra equipment and cabling is less likely to be bothersome.</blockquote>

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This document was written and formatted and (CC) by dex Otaku [Derek Gunnlaugson] for MDCF on 2006-05-11. The contents of this post, though not the entire thread containing it, are the intellectual property of dex Otaku. Redistribution or quotation are allowed under the rules of Creative Commons: attribution is manditory. Commercial re-use of any kind is prohibited without the express written consent of the author.

Someone might want to sticky this, after reviewing it for errors.

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B) nice job!

I like the little disclaimer at the end. :)

But seriously, well done! Should defintely be sticky/pinned!

Edited by raintheory

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Already did it. Thanks again, dex..:)

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Good job. There are a few links that need to be fixed though (from 'Why should I use manual levels instead of AGC?' onwards).

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There are a few grammos in there, but as for the links - I have the original [text] document open here, as it was saved immediately before pasting into the "post thread" page here .. and all the links are correct in that document. The board appears to have added "forums.minidisc.org" to the beginning of all of them for some reason .. though wait .. the first few links were done with BBCode, the rest were straight HTML .. which means that either the board or the server is adding that to each link there as html. Is that canonical linking? I can't remember.

Content addition:

When I'm talking about mic sens settings up there [aka preamp gain settings] and suggesting what low and high sens can be used for .. here's a note on that .. when I say "low sens" is good for conversation, I mean .. between a few people who are a metre or two away. It's important to remember that distance from the source is an important part of things. Trying to pick up conversation [or, say, a lecture] with someone who's speaking more than a couple of metres away from you is a job better suited to "high sens" mode, for instance.

Common sense comes heavily into play with so many of these things. There are few constants involved in recording; every venue/location is different, every source or subject has its own characteristics. It pays to know a bit about the parts that remain the same [like how to operate your recorder, how sensitive your mics are and what the meters read at a certain perceived loudness...]

There are a few grammos in there, but as for the links - I have the original [text] document open here, as it was saved immediately before pasting into the "post thread" page here .. and all the links are correct in that document. The board appears to have added "forums.minidisc.org" to the beginning of all of them for some reason .. though wait .. the first few links were done with BBCode, the rest were straight HTML .. which means that either the board or the server is adding that to each link there as html. Is that canonical linking? I can't remember.

Content addition:

When I'm talking about mic sens settings up there [aka preamp gain settings] and suggesting what low and high sens can be used for .. here's a note on that .. when I say "low sens" is good for conversation, I mean .. between a few people who are a metre or two away. It's important to remember that distance from the source is an important part of things. Trying to pick up conversation [or, say, a lecture] with someone who's speaking more than a couple of metres away from you is a job better suited to "high sens" mode, for instance.

Common sense comes heavily into play with so many of these things. There are few constants involved in recording; every venue/location is different, every source or subject has its own characteristics. It pays to know a bit about the parts that remain the same [like how to operate your recorder, how sensitive your mics are and what the meters read at a certain perceived loudness...]

RE: the links, someone else will have to repair them since I can't.

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Great addition Dex, as someone who has done a lot of recording, your points have much merit. Getting used to digital recording does take getting used to if you are used to analog, (reel to reel, etc), as they tended to be more forgiving of minor level mishaps, and most had good level guages and easily adjustable level controls to correct mistakes on the go.

Keep it up,

Bob

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thanks dex... this is really one of the main reasons I keep coming back to MDCF and keep pointing ppl this way! I'm still learning (and my recordings are still open for improvement) but thanks to you, greenmachine and A440 (mainly) at least my mistakes now have very little to do with 'abusing' my equipment (darn that sounded rancy... sorry :P )

as a request for a future contribution (perhaps for a coop between you three rec-gurus :lol: ) I would be interested in a simple lesson on auditive/visual 'reading' of a venues accoutsics (i.e. what's most important to look/listen for when choosing a spot) and perhaps some guidelines for a couple of 'frequent cases' (i.e. the small and always too loud club, the open air festival,...)

yes, I know there isn't any way to tell something accurate about a location you don't know, but some generalities like when to aim for a bit of distance and a nice stereo effect, when to 'hug the speakers' and go for a clean but mono sound, etc... just some case descriptions from your experience so we slowly learn to look/isten for the right things at our local venues...

still, only a suggestion and this contribution is already very highly appreciated!

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Dex, this is great as always. Someday we'll have to figure out why your in-ear binaurals don't overload the preamp while my BMC-2 do. But in the meantime, I hope this helps people.

I would be interested in a simple lesson on auditive/visual 'reading' of a venues accoutsics (i.e. what's most important to look/listen for when choosing a spot) and perhaps some guidelines for a couple of 'frequent cases' (i.e. the small and always too loud club, the open air festival,...)

I fear you're asking the impossible, Volta. Absolutely the only thing that matters in a venue is what your ears tell you, not your eyes. Visual cues only go so far. In a small club, relocating by just a foot or two can suddenly clear up the mix.

My basic move is to get near the soundboard. Presumably they are mixing what they think we should hear. But again, those few feet between me and the sound engineer can make a big difference.

I don't crush up close to the stage because usually the PA is pointed over your head and what you hear is a messy mix of stage amps and stage monitors.

Other cues: I try to stay away from hard-surfaced walls because they reflect sound harshly--unless I need them to provide some highs. I try to stay out from under overhanging things, like balconies, because the sound bounces around underneath, usually adding bass and muffling everything else. Being up in the balcony usually isn't as good as being down on the floor--though in some venues, front row balcony is ideal for both sight and sound.

At outdoor festivals you're usually better off pointing the mics at the speaker tower because that's what you're listening to, not the stage sound. But at an open air festival the biggest consideration is to get away from people who are talking.

Just think about basic acoustics. Hard flat surfaces are reflective. Complex surfaces, like statuary or plaster ornaments (in beautiful old concert halls) diffuse the sound in good ways. Soft ones (curtains, clothing, people) are absorptive. That can help you guess where good sound will be. But the only way to know is to stand in a bunch of spots and close your eyes. In New York, I know where the sweet spots are in most of the places I go to. But I couldn't have predicted most of them.

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see... A440, what I asked wasn't really impossible :P as I think his is a pretty good answer already :lol:

auditive/visual 'reading' of a venues accoutsics (i.e. what's most important to look/listen for when choosing a spot)
as you se, I did learn already that simply looking isn't enough... but as you explained a bunch of things that will affect the sound, can already be seen quite clearly (ornaments, curtains, balconies :lol:)

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Addenda:

While testing my equipment today [had a problem with my mics which has been fixed - but it's time for new mics] I took a moment to actually listen critically to the difference between AGC's STANDARD and LOUDMUSIC settings.

STANDARD mode: has no hold and a release which lasts about 2-2.5secs to return levels to normal.

LOUDMUSIC: has an infinite hold. I listened for about 90 seconds and it still hadn't released since the intitial tap I made on the mic.

Now .. I never thought about this before. STANDARD mode might be compressing/limiting .. and LOUDMUSIC might be just setting the levels according to the loudest peak that occurs - without compression. Intriguing.

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I took a moment to actually listen critically to the difference between AGC's STANDARD and LOUDMUSIC settings.

STANDARD mode: has no hold and a release which lasts about 2-2.5secs to return levels to normal.

LOUDMUSIC: has an infinite hold. I listened for about 90 seconds and it still hadn't released since the intitial tap I made on the mic.

Now .. I never thought about this before. STANDARD mode might be compressing/limiting .. and LOUDMUSIC might be just setting the levels according to the loudest peak that occurs - without compression. Intriguing.

I saw your post earlier today and have done a similar test on my RH10, PCM mode. I've examined the waveforms using Audacity. I used the noise of my computer's fans for the steady low level background noise, peaking at around -30dB, and beat a wooden tray to make the loud noise.

I'll agree that in Standard AGC mode, after a sustained loud noise it appears to take about 2.5 seconds to return from a low gain level. In LoudMusic mode, my RH10 recovers in about 20-25 seconds. It doesn't have the infinite hold that you found.

What I noticed in my initial tests was that in both modes, a single handclap was simply clipped just below Peak level in either case, and there was no ducking of the AGC. In Manual mode, a handclap would cause overload and the red overload marker would be set in Audacity. As these results, for the Loud Music mode are different to yours, I may see what happens with my NH900.

Thanks for posting - I now understand the action of the AGC slightly better than before.

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I used the noise of my computer's fans for the steady low level background noise, peaking at around -30dB, and beat a wooden tray to make the loud noise.

heh, sounds like something I would do for some of my experimental music.. ;)

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I'll agree that in Standard AGC mode, after a sustained loud noise it appears to take about 2.5 seconds to return from a low gain level. In LoudMusic mode, my RH10 recovers in about 20-25 seconds. It doesn't have the infinite hold that you found.

What I noticed in my initial tests was that in both modes, a single handclap was simply clipped just below Peak level in either case, and there was no ducking of the AGC. In Manual mode, a handclap would cause overload and the red overload marker would be set in Audacity. As these results, for the Loud Music mode are different to yours, I may see what happens with my NH900.

It's possible that this is a difference between gen1 and gen2; I didn't test my RH10, only the NH700. And it might still not be infinite hold - it just lasted a looooong time .. 20-25secs is still a long time in itself.

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I've now repeated the Standard/LoudMusic AGC test on my NH900. In the Standard AGC mode, there's hardly any difference between my NH900 and my RH10.

However using the NH900, in LoudMusic AGC mode, immediately after the cessation of the loud noise, the gain is at a very low level indeed. In the next 1/2 second, the ambient noise quickly increases to a level about 4 to 6 dB below the level it was before the loud noise. After approx 2.5 seconds, the ambient noise then decreases to ~ -60dB, and then from here, there's a slow return to the normal level of the amnbient noise, similar to how my RH10 behaved, i.e. taking about 10 or more seconds.

In LoudMusic AGC mode, the variations in level are audible and puzzling, and as I've described from the waveforms displayed in Audacity, all is "explained". It seems as though there's more than one time constant fighting another.

In all cases I fed a mono signal to both Left and Right, and both channels behave similarly, and with AGC in operation the red overload indicator wasn't triggered.

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Nice work, Malcolm. Thanks for the additional info.

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when recording fm radio or something through line-in should I use AGC or manual levels?

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when recording fm radio or something through line-in should I use AGC or manual levels?

I'd always tend to record FM radio at a fixed level, set high but so that distortion or clipping doesn't occur, but I think it could depend on how you intend to listen to the result. If you want background music and the source level is varying, using ACG might just give you a more useful result. I do find the AGC on Sony's Minidisc recorders is quite powerful, and noise pumping quickly becomes obvious. Equally, I've found that recording manually and a little down from max level, and then using the "Amplify" function in Audacity, I can bring levels up after the recording session, without (too) obviously raising the noise level.

Recently I've recorded some recitals where the spoken introduction between pieces was at a very low level (no PA amplification versus amplified instruments), and 20-30dB gain in Audacity was useful to make the intros audible! (Here, quality definitely wasn't an issue!!)

HTH

Edited by Malcolm Stewart

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as I believe the outgoing signal from FM radio-stations (so your incoming signal) is quite 'stable' in loudness ...well, that's what I've learned from one guy who works in radio and from my personal listening experience (how often do you find yourself changing the volume of your radio?)

with this in mind I don't reckon it really makes a difference wether you use AGC or manual... manual will record what you hear exactly as you heard it (but it miht need amplification afterwards, requires going through more menus - except with the RH1 of course - and probably is a bit of overkill for radio IMHO)

AGC could theoretically cause swooshing sounds with sudden adjustments, but I guess the signal from FM is almost kept stable at a certain volume (I hardly ever have to 'listen closer to hear what they are saying' or to 'rush and turn it down as a loud track starts') so AGC shouldn't really be working most of the time... you could use it (on 'louder music' setting or what's it called) as some sort of buffer... it will only work to prevent clipping but almost not for anything else I guess

best thing is to try both methods and see what pleases you most (heck, it's FM-radio, this must give you the opportuniy to try it before you have to do it 'for real'... at least this should be much easier than with concert-taping... how often do you have some less interesting/free concerts you can go and tape just for testing? :lol: )

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This is another matter of personal preference as well depending on what kind of recording you do – a fixed-gain preamp might be perfect for certain situations, while being generally less versatile.

I would personally recommend using an external preamp for things such as nature, ambient, environmental, or acoustic music recording – generally all things where detail and dynamic range are of importance from the get-go. The potential lower noisefloor, probable higher gain and headroom of the preamp will make the investment worth it; also, since many of these types of recordings are made rather deliberately, the inconvenience of the extra equipment and cabling is less likely to be bothersome.</blockquote>

Thanks for your lecture, very interesting. I'm a beginner at recording. I have a MZRH1 and I need to record speech as cleanly as possible for dvd. My first experiment last weekend with a sanken cos-11 lavalier and a beachtek DXA4 did not go as well as I'd expected. I didn't and may not be able to modify the ambient noise level much as I will be recording old folks stories. The MZRHI was set to low mic sens, PCM and the AVL was off. The beachtek's levels were set to high. The recording was very low volume much to my dismay. What might have caused this? I was able to use gain in the video editing program which worked alright, however an audio tech I spoke to said it should be the other way around, high volume recording brought down to around 30db's in the dvd.

Dan Brocket describes the Beachtek as a routing or impedance conversion device (link and file below), is that then a pre-amp or something else? Since the sanken mic has an XLR connection, I'll need something between it and the minidic, what would you recommend? A different mic perhaps?

http://www.kenstone.net/fcp_homepage/location_sound.html

[attachmentid=2078]

post-44821-1163071227_thumb.jpg

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I'm not familiar with either your mic or the Beachtek, so this may be off-base, but...

It looks like the output of the Beachtek can be at either mic level (which is lower because there is a preamp behind the mic jack) or line level (which is higher, no preamp).

Is it possible you were outputting mic level into the line jack (white)? That would give you a quiet recording.

Try some combinations. Try setting the Beachtek at Mic level and going through the (red) mic jack at High and Low sensitivity. And try setting the Beachtek at Line level and going through the Line-in Jack (white).

Looking at the specifications, the Sanken needs phantom power--48V--unless it is the COS-11BP. The minidisc only provides 2V (which is called plug-in power and confuses people), and a battery module would provide 9V, neither of which is enough. I don't know if the Beachtek provides phantom power, but it doesn't appear to do so. So while that looks like a fantastic mic, you might need to invest in a phantom power module to use it.

To my way of thinking means it you would be better off just getting another mic--one with a stereo miniplug that goes into the mic jack. If you are in the United States, try http://www.soundprofessionals.com or http://www.microphonemadness.com or http://www.bhphoto.com for lapel mics.

For the cleanest sound, you can go mic--preamp--Line-in. But you should see if you get good enough quality through the mic jack. The minidisc's own preamp is pretty good.

Microphone recording should not be complicated with your RH1, and my motto is that the fewer boxes you use, the fewer things can go wrong.

EDIT: I just looked at phantom power boxes, and the AC ones are about $50, so that's not a big investment. Since the lapel mic you have is probably very high quality--it had better be at $400--then maybe you should get a phantom power module for it instead.

Edited by A440

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Im pretty much a noob at all of this. Ive the MZ-M100 and a really nice mic. but as of now I am recording out of Mic In..... ive heard of battery boxes and what not, and not really sure where to start looking. someone also told me I could make my own. but either way. is that all I need to start recording out of line in? and where can I start looking for one? minidisco.com is where I got my stuff but they dont have anything on battery boxes.

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Here's the one I use, or look at Greenmachine's build-it-yourself instructions.

http://www.microphonemadness.com/products/mmcbmminminc.htm

Mic-->battery module-->Line-in. You will get much better recordings by setting Manual Level under REC SET.

Just don't forget to unplug the mic from the module after the recording, because the battery runs while the mic is plugged in.

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Hi,

Can I turn a portable headphone amp (Super Mini) into a mic preamp, if at all? I am on a dummy side, so if it's too complicated I'll probably skip it.

FYI, I've got Reactive Sounds DT-1 - Delta Stereo Microphone. Is it decent quality?

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This was a very well worded presentation sir .

"pumping" is actually the term that is used in compression dynamics.

Turn this into a PDF and put it up for download for the people who dont know how to set levels . Very nice disortation indeed.

TC

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