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Gear / Technical Help => Ask The Tapers => Topic started by: matsa on March 19, 2011, 09:54:40 PM
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Always trying to improve my rig, but I'm currently stuck. Can't figure out which part(s) to replace to get better results.
Ok, I record my vocal ensemble and some classical music. Noise must be kept low as possible, since the music is rather silent compared to rock shows.
Also, I like a rig that is minimal as well as easy+quick to set up (=few units and single point mic preferred). I could add a phantom supply, but would like to avoid it (=battery powered mics preferred unless phantom and pre are combined).
Today, I use: Rode NT4 -> Denecke AD20 -> iriver 140 (battery&memory card modded).
I'm happy with the current rig, but I kinda like buing new stuff. What would you suggest for my improving the above rig?
Thanks!
/Matsa
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What about moving to a 24 bit recorder? The Sony M10 seems to be the weapon of choice for many around here.
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Naiant LittleBox>Sony M10
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Naiant LittleBox>Sony M10
for sure.
http://taperssection.com/index.php?topic=128321.0
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Does your current rig not meet you needs or do you just have the itch to spend money? I understand the latter completely. You have made no mention of $$$, so I will assume the sky's the limit.
If you are trying to simplify your rig going to an all-in-one recorder with built-in preamps seems to be the way to go. If you want quiet the recommendations are the usual suspects: Sound Devices 722 or 702, Fostex FR-2LE, Marantz 661. These are the ones I have used and they are all capable of making fine recordings. I prefer the Sound Devices boxes, but of course they are the most expensive, by about a factor of 3. I am sure there are other acceptable recorders out there that others would recommend, but the ones listed above I have used.
New, good, quiet, single-point stereo mics are scarce these days. Lots of inexpensive low-end stuff is geared towards audio for video use and is not very good. On the other extreme, Neumann and Schoeps make a couple of stereo models between them which I am sure are fine mics but I have never used them.
Three stereo mics I have used and can recommend are (in order of increasing price) Busman BA-SR1, Busman BSCS-L and the Nevaton MC48. The BA-SR1 is a very nice sounding (especially with the Lundahl transformers) stereo ribbon which sounds great on classical or jazz vocal groups. Nice on strings too. However, it is a ribbon so it is somewhat delicate and has low output so it need lots of quiet preamp gain. The BSCS-L is a good sounding, moderately quiet, versatile LDC stereo mic. My biggest complaint with the BSCS-L is it's big and heavy, but some like 'em big. Busman Audio is a member here. http://www.busmanaudio.com/index.html (http://www.busmanaudio.com/index.html)
The LDC Nevaton MC48 is neutral sounding and quiet. Not as versatile as the BSCS-L, but quieter, smaller and lighter. Does the MC48 sound better than the Busman mics? That's a matter of taste and the music you're recording. Nevaton is sold by a member here: http://bigpurpledog.com/mc-48 (http://bigpurpledog.com/mc-48)
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Great tips! Thanks a bunch all!
@notlance: you're so right! Although I'm not unhappy w my current rig, it could always sound a little bit nicer. Also, buying new, better stuff is great!
For starters I'll look into the LittleBox.
Please continue to share your advice!
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littlebox > M-10
see my pics here:
http://taperssection.com/index.php?topic=143728.0 (http://taperssection.com/index.php?topic=143728.0)
--Ian
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also in the future upgrading the mic- not that the NT4 is a bad mic- but there's a lot of other "flavors" out there! While I still have my NT5's I hardly ever use them and always turn to my Milab VM44 links
but at like 4x + the price of the NT5's it took me a while to get there.
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And I forgot to mention, get the LittleBox w/ OUTPUT Transformers
Here are a few of my MBHO Hypers>LittleBox[With OUTPUT Transformers]
Trey 2/25/2011
http://bt.etree.org/details.php?id=542790&hit=1
Dead Kenny G's 3/12/2011
http://bt.etree.org/details.php?id=543244&hit=1
Keller Williams 3/19/2011
http://www.archive.org/details/kw2011-03-19.ka500.603a.lb.r09.flac16
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Today, I use: Rode NT4 -> Denecke AD20 -> iriver 140 (battery&memory card modded).
I'm happy with the current rig, but I kinda like buing new stuff. What would you suggest for my improving the above rig?
^^^ never heard that around here before.lol
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Each aspect of your gear chain could use an "upgrade" if you're talking about achieving the best specs, of course... But this isn't as cut and dried as it first appears, I guess.
The AD20 is a fine AD converter, so you're basically just using the iRiver as a bit bucket. The iRiver is 16bit and therefore not IDEAL, but I'm not so sure the 16 vs. 24bit difference is THE critical aspect of your rig. In some ways, if you're being encouraged to switch the mic to phantom and then run it all analog through a littlebox (a great inexpensive preamp) into the iRiver, I could see that maybe being worse than the nice digital signal path you have now. So, as pointed out, you'd need a new deck, too. But that still leaves an analog signal path where you used to have a digital one.
I would think about the potential performance improvements from running that Rode mic off of phantom instead of batteries. As everyone here will tell you, mics are the most important aspect of a rig. I would think that either getting a better, phantom-powered single-point stereo mic, or strongly considering a phantom powered stereo pair, might be better. A stereo pair offers more placement options and greater upgradeability in terms of using different capsules for different polar patterns.
Having a budget here would help, as that would clarify things in terms of what your realistic options are.
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Again, thanks for all the input! Much appreciated.
Yup, the iRiver is a bit bucket. However, I recall reading that the truncation from 20 to 16 bits (AD20 to iRiver) may actually be beneficial from a noise perspective, although it's only 16 bits.
Keep in mind that I record "quiet" sound sources, so noise/hiss is the big issue.
I deliberately didn't give a budget in order to get all kinds of suggestions raining, and I'm still hesitant since is may impede creativity. But since it is requested; for a mic upgrade, I could go to 1k$. Possibly more. But, out of convenience, I feel I must stick to stereo mics (battery power is a +) - my former rig had two mics, but I constantly found myself trying to opt that away by clever single point holders to get an ultra quick set-up of the equipment.
BTW, from a noise and hiss POV, how does integrated pre+recorder units like the Fostex FR-2LE or the Marantz 661 compare to, for example, an AD20->iRiver (or similar) combo?
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Nature recordists (those who record animal and natural sounds) are very concerned with recorder and microphone noise, so they are a good source of information on the subject. Here is a table from a nature recording site listing the noise specs of all the recorders you mentioned:
http://www.avisoft.com/recordertests.htm (http://www.avisoft.com/recordertests.htm)
This is also worthwhile reading:
http://www.avisoft.com/tutorial_mic_recorder.htm (http://www.avisoft.com/tutorial_mic_recorder.htm)
http://www.rane.com/note148.html (http://www.rane.com/note148.html)
I record mostly classical music in quiet environments at 24bits/44.1 kHz and the room has always been noisier than my mic and recorder combination. I've used the 661, FR-2LE, and the 722 recorders, but I have much more experience with the Sound Devices recorders. However, I believe that if you record at 24 bits, noise will not be an issue with any of these recorders.
I do not understand how truncation from 20 bits to 16 bits could decrease the noise; digital systems don't work like that.
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I was looking for that avisoft link! Thanks for posting.
So as you can see the fr2LE is pretty quiet when compared to all those other recorders and pretty affordable.
Since you're interested in single point stereo, have you considered anything like the core sound tetramic? Only a couple people on this forum have it but it may be some interest to you. This would be an upgrade to at least a four channel recorder to accommodate all four channels, but you could put this on a stand and hit record knowing you can do all adjustments later with software. The self noise may be a little concern at 19 dBA per capsule but how is self noise measured anyway. ;)
of course, this mic requires four channels and phantom power.
http://www.core-sound.com/TetraMic/1.php
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Thanks again!
Having read the posted links carefully, I find that the BP4025(which came to my attention)->FR-2LE seems like a nice combo as long as the sound source is not loud.
However, I read both good and bad about the AT BP4025 performance.
Does anyone have experience of the 4025, or even the combo 4025+FR-2LE for recording e.g. a choir?
Do you think that I can avoid a separate preamp unit by choosing the FR-2LE, or am I better of with e.g. a littlebox+a cheaper recorder?
Your comments are as always much appreciated!
P.S.
A few years ago I was very excited about the tetramic, but came to the conclusion that for me, a dummyhead could deliver better results. That project is still on hold.
I haven't given the truncation influence on noise much thought, but I liked the idea since I was about to set up the AD20->iRiver combo. :)
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LB>M10 ;)
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yeah, I truly believe that going 24-bit is the best thing you can do, for a start. It gives you a lot more headroom for boosting and editing later, so that you can run a little low and still get a good recording.
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yeah, I truly believe that going 24-bit is the best thing you can do, for a start. It gives you a lot more headroom for boosting and editing later, so that you can run a little low and still get a good recording.
I agree
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Today, I use: Rode NT4 -> Denecke AD20 -> iriver 140 (battery&memory card modded).I'm happy with the current rig, but I kinda like buing new stuff. What would you suggest for my improving the above rig?^^^ never heard that around here before.lol
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LB>M10 ;)
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(http://i900.photobucket.com/albums/ac210/OtheroneK/AT%20Mics/P1100793.jpg)
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;D
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LB>M10 ;)
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(http://i900.photobucket.com/albums/ac210/OtheroneK/AT%20Mics/P1100793.jpg)
=
;D
Once again, agreed ;D
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Yep, message received. That is a killer combo. :)
But... I have located a couple of used fr-2le in my area, and I think I'll get one.
This way I also go from two boxes to one.
At a later point I'll look into the stereo mics from LineAudio design (ST6Li, http://www.lineaudio.se/linemic.htm ) as well as the BP4025.
Thanks again!
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Interesting mics... and very inexpensive at ~$160USD each..
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If you want low self noise out of a mic, Large Diaphragm Condensers are generally quieter than Small Diaphragm condensers, per the laws of physics. I'm not saying all LD's are quieter than all SD's without exception, but within a similar price range, it is frequently true. The flip side is that the mic tends to be bigger and heavier. I ran an LSD2 > PS2 / AD20 > H120 a few times, and it sounded pretty nice. Yes, that's a phantom powered mic, and it's big and heavy. The AD-20 is generally regarded as a pretty quiet pre, and very easy on batteries.
You mentioned the AT4025... I've never heard one, but if it's similar to the AT825, that was a decent mic, but I wouldn't rank it as a huge upgrade over the NT4.
Most of the improvements in a rig are made on that analog side of the chain... mics + preamp... right up to the A/D converter, and sometimes including the A/D. Having a 24bit or 16bit "bit bucket" is much more subtle than most people think. Breaking down the sample into finer resolution isn't nearly as important as having a good signal to sample.
The other big thing is that mics and preamps have "flavor" which is difficult to describe. Some combos go together great (like peanut butter and jelly), and some are terrible together (like oil and water). Predicting the good combos versus the bad combos isn't always easy. If you spend a lot of time on archive.org, you will find certain combos are used a lot.... So what does that tell you? People tried it and they liked it, so they kept it and ran it for years. Pick some other combo at random, and you won't find many sources... maybe no one has tried it, or maybe they did try it, it sucked, and they bought something else after a couple of shows... Darwinian selection. That's the advantage to finding a "tried and true" combo. If you let people like us throw out random suggestions and put those pieces together, it's anybody's guess how they might sound, or if you will find that pleasing. But we do get to learn if a combo works or not, and you get to pay for it.
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^^ QFT
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^^ QFT
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Most of the improvements in a rig are made on that analog side of the chain... mics + preamp... right up to the A/D converter, and sometimes including the A/D. Having a 24bit or 16bit "bit bucket" is much more subtle than most people think. Breaking down the sample into finer resolution isn't nearly as important as having a good signal to sample.
(my bold above) Just picking nits in a spam bumped thread.. ;)
I agree in a practical sense with everything mentioned in Joe's post, but just want to clear up a common technical misconception in regards to the signifigance of bit depth and sampling rate. I think the phrase 'finer resolution' often misleads people who do not have a strong understanding of how PCM sampling works. As long as the bit depth used is sufficient to cover the dynamic range of the analog signal being recorded, increasing the bit depth from 16 to 24 bits does not mean storing smaller, finer, or more accurate incremental values of loudness within that range. Instead, increasing the bit depth increases the total number of recordable values, which then cover a proportionally larger range of signal levels. A higher bit rate is capable of recording a wider overall range of signal levels. That is advantageous to us as recordists because we do not know beforehand what the absolute signal levels will be, so extra unused room at the top and bottom of the recorded range can be especially useful, even if the extent of that increased range is far greater than need to reproduce the source signal with the same acuracy.
Similarly, a higher sampling rate does not record more information than a lower sampling rate does within the frequency range covered by the lower sampling rate, it extends the recorded frequency range to higher frequencies. Higher bit rates extend the overall range of dynamic levels.
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^^^ In layman's terms, does that not (possibly) mean:
"If recording a PA system with limited dynamic range, a lower bit depth (say, 16bit) and sampling rate (say, 44.1) is likely sufficient to capture the full existing frequency range"?
I'm not arguing the merits of what anyone does - I record at 24/48 myself "just to be sure" - but scientifically, isn't that the case? Unless of course PA systems have much higher quality than I think they do....
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"If recording a PA system with limited dynamic range, a lower bit depth (say, 16bit) and sampling rate (say, 44.1) is likely sufficient to capture the full existing frequency range"?
I think this conflates dynamic range and frequency range a little (at least as written). For a source with limited dynamic range, a lower bit depth should be sufficient to capture it (16 bits is ~ 96 dB dynamic range). The bit depth doesn't really have anything to do with the frequency range, though. That's where the sampling rate comes in. Basically, you can completely reconstruct the waveform for frequencies up to the Nyquist frequency (one half of the sampling frequency). So at 44.1, you are good up to a frequency of 22.05 kHz. At 48, you are good up to 24 kHz. Human hearing is generally quoted as going up to 20 kHz, so you could argue that any sampling frequency past 44.1 or 48 is unlikely to yield any audible benefit...
So, really, assuming I actually understand this stuff, the above quote should be split into two parts:
"If recording a PA system with limited dynamic range, a lower bit depth (say, 16bit) is likely sufficient to capture the full existing dynamic range" and
"If recording a PA system with limited frequency range, a lower sampling rate (say, 44.1) is likely sufficient to capture the full existing frequency range"
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My personal/practical take away is that the 16/44.1 PCM in itself is entirely sufficient as a reproduction format for any music source, including highly dynamic material with a full frequency response. Yet I feel there are real advantages for recording and processing at higher rates and wider bandwidths, within practical limits of diminishing returns. I've settled on recording raw files at 24/48 and the software I use for processing works at a further increased bit depth and somtimes oversamples internally for its calculations. Yet once all that is complete and the recording is optimized for playback, I don't feel the difference between 16/44.1 and higher rates is significant enough for me to be very concerned about.
In layman's terms, it's about sufficient working space and wiggle room.
Care for a few tortured analogies? My finished acoustic guitar sounds great and fits in a snug case which is the perfect size to hold it, a bigger case wouldn't make it sound any better, but there is no way I could build an equivalent guitar inside the case, I'd need much more workspace to do it right. Yet once completed, the small case is perfect and more efficient than storing the guitar in a dedicated room.
There is a precise amount of gas required to get your car from point A to point B, however the practical question in the real world is rather "am I sure I have enough gas to make it?" How close are you willing to cut it before running out of gas?
I brought this up only to correct what I think is a common misconception about a clear black and white technical aspect of how the format works. In practical 'real world' implementations, subjective perceptions and all kinds of confusing complications arise. A few- Yes a 44.1 sampling rate can theoretically record and reproduce signal frequencies of up to 22 kHz. Yet to do so the frequencies above 22kHz must be completely filtered out, and real world practical filters which don’t sound bad require some range over which to work, so rolling off starting at 18 or 20kHz might be a better engineering choice. Theoretically a 16bit PCM file has a dynamic capability of up to 96dB, but how many ’24 bit’ recorders that we use have an actual dynamic range capability of much more or even that much? Different implementations, chips, filters and circuits both measure different and sound subjectively different to people, but that’s the down and dirty practical engineering and subjective side of things.. which people argue endlessly.
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I really hate to interrupt this discussion
SmokinJoe: "[...] But we do get to learn if a combo works or not, and you get to pay for it."
Quite right. Said and done: Picked up a used, mint condition FR-2LE two weeks ago, but I've had too much on my plate to take the NT4->FR-2LE combo for a proper spin. Running on phantom, with supposedly more silent preamps should be an improvement. If not, I'll buy&try other mics.
Will let you know shortly.
Again, sorry to interrupt. Go on. :)
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Short answer: a pair of SDCs would provide dramatic improvements in your recordings, far more than any other piece of gear and/or a different single-point stereo mic.
Long answer:
I know you said you like gear that's minimal and easy & quick to set up*, but I strongly recommend considering a pair of SDCs. A modular system would prove ideal, allowing you to switch between cardioids or omnis (or even hypercardioids) for different recording environments, but even a pair of fixed cardioids would produce big improvements. Using a pair of mics would allow you to use a variety of coincident and near-coincident mic configurations, instead of the single fixed, coincident configuration of a single-point stereo mic.
The ability to change the included angle alone, even if still running a coincident configuration, would have a significant impact. But running near-coincident mic configurations would deliver even larger improvements. The primary benefits of running near-coincident configurations:
- Time-difference stereo. Coincident configurations provide intensity-difference stereo. Near-coincident configurations provide intensity- and time-difference stereo. Generally, and all else being equal (e.g. stereo recording angle), mic configurations with time-difference stereo produce what some might call a greater sense of space.
- Flexibility and control. Due to the ability to change not only included angle, but also spacing, you have far greater flexibility in controlling the sound of the recording. You can change the configuration to control the stereo recording angle and, to a significant degree, the relative levels of direct and reverberant sound captured by both mics. All else being equal (like stereo recording angle)...want to capture less reverberant (i.e. "room") sound? Try wider spacing with narrower included angle. Think the room sounds great and want to capture more of the "room" sound? Reduce spacing and increase the included angle.
* It's not much more hassle to set up a pair of mics, and IMO the benefits greatly outweigh the additional hassle. And there are ways you can streamline the setup, e.g. leave the mic bar and/or shockmounts on the stand. The AKG KM 235 stereo bar is great in this regard because it allows one to fold the bar towards the center of the stand to minimize the on-stand footprint of the shockmounts and mount bar for transport.
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^^^
Best advice yet, actually on topic addressing the original question!