April 27, 2007

Mics and Phantom Power (48V)

Here's a question we answered today about someone trying to use a condenser mic that has an XLR connection (a balanced, powered, shielded type of connection, the best you can use in a studio) and required phantom power, as most good mics do. Here's what we wrote, about the Digidesign Mbox 2 and the really impressive "WE WANT ONE" MXL 990 mic he has:

Just to be complete, the "Digidesign® Mbox® 2 Pro" - assuming you got the pro model, not the original, is an "audio/MIDI production system ... packing an impressive variety of connection options into a compact interface". I'd read about it before and it sounds like a great box for a lot less than other similar products.

The MXL 990 mic is a condenser, shock mount mic I've read good things about in Tape-Op Magazine, cardioid pattern with 30Hz-20KHz range (claimed) that requires a shock-mount. Similar to my Audio-Technica 4033 but with more on the bass end. A great mic from what I've read.

Anyway, it's mono, of course, and has what's called an XLR connection, and most importantly, requires "48V Phantom Power (+/- 4V)", so it's a phantom-powered mic.

So you'll need an XLR cable for the mic, and something that supplies phantom power. I plug my phantom powered mics directly into my Mackie mixer, which provides 6 phantom-powered XLR jacks for mics (you can turn the phantom power on/off.)

So the question is, does the Mbox 2 have XLR inputs, and looking at the photos it does, let's see... specs say "4 analog inputs (2 XLR/1/4” TRS combo jacks & 2 1/4” TRS jacks)" so you have 2 XLR jacks, so all you need, cable-wise, is a standard, shielded (!!) XLR-to-XLR cable.

Checking the specs further, the Mbox 2 offers "48V phantom power for condenser microphones", so you're good to go, you just need the XLR-to-XLR cable and if the phantom power is something that needs to be turned on, turn it on. As for output, the Mbox 2 offers 8 audio outputs, but they all look to be line-level outputs, so you'd need to run the output either to powered speakers, or to a pre-amp.

This Mbox 2 even offers a phono pre-amp with RCA hookups so you can hook up a turntable to it. Quite the unit, it covers everything.

My guess would be that if you have the right cable and the phantom power turned on, that your output needs to go to powered speakers/monitors (ones that have their own amps built-in, and provide better sound since the amps are designed specifically for the speakers/monitors) or you need to go to some kind of pre-amp from the output in the Mbox 2 and then plug your speakers into the pre-amp.

Should all work just fine. Bad cable is something to check for too, as always!

The two key things here are that many people don't understand that some mics can generate enough output to make line-level signals which usually don't need to be boosted, but really good mics, regardless of whether they're condenser, ribbon, PZM (piezo-electric), transducer, etc. usually need what's called "phantom power". The best analogy is the output from a turntable, which requires a "phono pre-amp" just to boost it to line-level output. Phantom power mics need a 48V boost to get them to line-level, and either you need a soundcard, mixing board, or some other pre-amp to supply that phantom power to boost the system.

The other thing is to make sure you have good cables, as many times people can waste a lot of effort when a bad cable is involved. Phantom mics are also very easy to fry so a bad cable can ruin it, and they should come with an "off" switch, and one should always turn off the mic before turning on the phantom power, except for PZM mics, or you can literally burn out the mic if there's a power spike when the phantom power is turned on.

Good mics are delicate things - learning to use them correct is very important. If there's anything to be understood about mics, that's Rule #1.

Posted by Wink Junior at 12:31 PM | Comments (0) | TrackBack

April 24, 2007

Converting Cassettes to CDRs

We recently fielded this question on AllExperts.com:

You can definitely find places that offer this service, such as our studio. http://sound-o-mat.com/transfers/ for more info.

If you want to do it yourself, you basically need a soundcard that you can hook the line level of a cassette deck up to, which usually means a cable to convert the two red/white RCA plugs to a single 2mm mini-plug that goes into most soundcards, then you'll need sound-recording and editing software. There's a lot on the market and some is quite expensive, such as Sony's SoundForge or Adobe Audition, but we recommend Audacity, which runs on Windows, Mac and Linux, and will do most of what you need.

You then need to play the cassettes, record them, edit the files, clean up the tape hiss and any other noise or artifacts, then you can burn them to CDRs. There are also many popular CDR burning programs, such as Nero and CDRWIN. It's up to you.

While this deals with vinyl to CDRs, you can also read this article, which will explain some of the details involved in converting any format to digital and then burning CDRs:

How to Convert Vinyl to CDs

Posted by Wink Junior at 08:27 PM | Comments (0) | TrackBack

April 18, 2007

Surround Sound: Not for Radio or most CDs

We recently fielded a long list of questions from a gentleman who wanted to buy an "AM/FM Radio with a CD Player either built-in or separate and a Surround Sound system for his new house, which had capped access built-in."

It was an interesting question if only because it was so long and confusing and contained a lot of information, but none of it was relevant to setting up a proper Surround Sound environment.

The most obvious problem was these "capped access" things that he kept mentioning, without bothering to say whether the access panels (?) or mount-boxes (?) had some kind of plug: co-ax, RCA cable plugs, or hook-ups for bare wires. Without this info, the detailed list of all the plugs and how one was for a sub-woofer (which he decided he didn't need) was pretty much just wasted time.

But the real problem was that although this person came across as fairly intelligent, or at least able to compose a question with proper grammar and spelling, he completely misunderstood what Surround Sound is. Hence this posting.

The short version is that Surround Sound is a way to take four, six, or eight channels (the latter two known as 5.1 and 7.1, where the ".1" part refers to a channel dedicated specifically to a sub-woofer and nothing else) and have the sound come out of multiple speakers, usually two or four on the floor and two or four "Satellite" speakers that you hang from the wall and which tend to handle upper frequency sounds only (which are very directional; the bass from a sub-woofer is non-directional) and get the effects of being, well, "surrounded" by sound as in the movie theaters that use Dolby or THX systems.

So what was the problem? It's that you need to have recordings that are actually encoded in some kind of Surround Sound format for this to actually "work", other than to just pump sound out of more than one or two speakers. And guess what? AM/FM radio is mono, not even stereo, and just about every CD manufactured is stereo (interestingly enough, due to the Sony/Philips CD specs, there's no way to record a mono CD, which is too bad, as it would double the amount of recording time, and is a feature Sony's MiniDisc standard supports.)

The earliest version of this were vinyl records recorded using "Quad Recording", which I'll be writing more about in a series of posts about the evolution of Surround Sound.

I'll write more later about Dolby's "Pro Logic" encoding, which was supposed to supply four channels as well, and did so slightly better but still nowhere near useful, and what Surround Sound is today. But back to the original point of this post: Surround Sound is for media that is encoded to use it, and that currently consists of maybe a handful of audio DVDs and about half the movies filmed in the last 10-15 years on DVD that use one of the Dolby multi-channel systems.

To hook up a mono AM/FM radio, or a stereo CD player, to more than two speakers is just a waste of time, money and effort, unless the speakers are in different places in the house or such (I have sound wired to four different rooms in mine.) But you're just not going to get any "surround" effect from a mono or stereo signal, period. It's that simple. If you want to build a Home Theater setup and buy DVDs that support Surround Sound, go for it, but to ask this about any kind of music media at this point is completely senseless. Unfortunately, that's what I had to explain to the guy, and I suspect he's just going to get pissed-off thinking I'm talking down to him or just because he didn't get the answer(s) he wanted. Which cracks me up - we're always amazed at the studio when we're working with clients or fielding questions, and you give them an honest, real answer, and just because it's not what they want to hear, they get angry. Fortunately, we've no problem just booting those people or refusing to work with them (or answer their questions.)

So at some point I will try to write up an interesting history of the evolution of Surround Sound, or more likely just find one on the Web and post a link to it, but in the meanwhile, at least know this about Surround Sound: if the media you're playing isn't encoded for it and carrying all the extra channels of audio, it's a moot point and a waste of effort. It's really that simple.

Posted by Wink Junior at 02:23 PM | Comments (0) | TrackBack

April 13, 2007

White Noise and Radio Sound on Stereo Speakers

This seems like a somewhat obvious question, but it's really not. Most people don't realize just how much "signal noise" there is going through the airwaves around their home: cell/mobile phones, CRTs and anything with an unshielded power supply, AM/FM radio signals, and a laundry list of others. Recently someone asked:

"Hoping you have an idea of what's going on here. I am getting background white noise along with a very low volume radio sound through my stereo speakers. This happens regardless of what component is selected,e.g. AUX, CD, Phono."

The only idea as to why the Bose system is picking up radio signals as well as white noise, likely from the wiring or radio signal interference, or perhaps internally generated, is the design of the Bose system. It could be that the Bose speakers aren't properly designed or shielded and are picking this stuff up. The other possibility is that the cables you're using are sensitive enough to pick up such signals and aren't shielded.

You don't mention the resistance of the JVCs vs. the Bose (8 ohms?) but my guess, without having that info, is that the Bose run at a lower resistance (4 or even 2 ohms instead of 8 ohms) which makes them more suseptible to picking up radio interfere, and a good set of shielded cables from your receiver to the speakers would help. To save yourself $$$, find the Bose unit that is the "noisiest" and try buying enough shielded speaker cable to plug that one in, and see if it goes away. If it does, and you want to keep the Bose system, then you can then go buy enough shielded cable for all of them.

That's about the best I can do with limited information. Do remember that it could be a problem with the Bose system itself, so if it's new you might want to return them and try another, as it might be a manufacturing defect, so get them back while they're still under warranty if you can.

Posted by Wink Junior at 09:36 AM | Comments (0) | TrackBack

April 12, 2007

Amps, Power, Speakers and Resistance (ohms) Explained

Ohms are a measurement of the amount of electrical resistance that the cables and circuitry of any electric signal carrying audio gear (or any electronics for that matter) produces. The lower the number, the more resistance, and the more overall power (watts) that the gear can support and is required.

You can't tell what an amp is running, since it switches automatically. It is dictated by the speakers you hook up. You need to check the rating on those, and if they're 8 ohms rated (most common) then that's what your amp will be powered at, if they're 4 ohms, it will be more wattage, etc.

If you have subwoofers that are 4 ohms, then that's what the amp will run them at, 600 watts RMS, probably around 1000 watts peak. You cannot "switch" or "change" the resistance: it is dictated by the design of the sub-woofer itself.

You're also working under a misconception - just because something is rated as requiring less ohms, which means your amp can use less power to run it. It will not give you "more power". The amp simply runs at whatever wattage is necessary to drive the speakers or woofers, and changing them will make no difference. If you want louder, all you can do is buy a more powerful amp, and make sure your speakers and woofer can handle that power without blowing out.

Posted by Wink Junior at 10:51 AM | Comments (0) | TrackBack

April 11, 2007

Mistakes in Mixing (Keep the Setup Simple)

Someone recently wrote about how they had mixed down a recording on a Tascam 2488 (a very popular and well-built digital recorder) and everything sounded fine, until they burned a CD of the mix-down (which the Tascam does) and the results sounded awful.

Matt wrote:

"The problem is that after pre-mastering a song and burning it onto a CD-RW (using the studios CD Writer) the final product sounds nothing like the mix itself.

Example: The guitars and bass sound fine... However, the drums never sound the same. There is no kick drum sound on the CD... But, the kick sound is VERY present when listening to the tracks through the studio monitors, and headphones."

We replied:

We believe you've run into a common problem with the recording process from mixdown to production to post-production and finally with mastering. Namely, what you monitor the mixes with can really make things sound a lot different, often better than they really are.

For example, for a second set of monitors, we made the mistake of buying the Alesis M1 powered monitors for our studio at The Sound-O-Mat (http://sound-o-mat.com). We read all the Alesis literature and studied the graphs and they were supposed to be the most "flat" (e.g. adding or removing any freq. ranges or "color" to the sound) on the market at the time. Big mistake. We should have waited to read the reviews, because it turns out they're extremely bass-heavy. They sound great on my DJ rig, which is where we use them now, but anything I mixed down in the studio sounded really flat on the bass and kick & tom drums.

The reason I tell you this story is that they're probably recorded OK, but you're listening to them from the Tascam 2488, I assume through the audio output, but it could very well be what you're listening to them with that made them sound good when they were recorded and now played back, but ultimately the source material itself isn't really that good. It could be the monitors you're using and/or the headphones (although it's great to hear you're using both - a mixdown should never be down with just one or the other.)

So that would be my guess. The combo of what you've recorded with the Tascam 2488 (good unit!) run through the Yamaha mixing console (you didn't mention which one) sounds great when they finally come out of that setup to monitors or headphones, but the CD is what the tracks really sound like, and they don't sound good on the players or stereo(s) you've tried (which you also didn't specify.)

This happens all the time. And it's very smart of you to have burned the CD and tested it that way. I personally believe you can't listen to enough mixes in enough ways (including from another room, if possible) to make sure everything sounds right.

So, how to fix it. Well, you may have to re-record the drums. That's the worst-case scenario, and I hate to be the bearer of bad tidings, but that might end up being the case. If so, consider it a lesson learned, and figure out some way to monitor the drums as directly as possible. For example, if you're mic'ing them (you didn't mention if they were real drums, samples, or synth-based) but with that assumption, if you can mic them directly to the Yamaha console and listen to them on monitors and headphones directly from the console (use the headphone out on it, and direct the signal from the channels with the drum mix to them with all other channels canceled, aka turned off) and keep the Tascam out of the recording chain, until you're getting the sounds you want. Only then would I add the Tascam back into the equation and record the drum tracks.

Now, as for fixing them, there are ways to do it, that require post-production work, using specialized audio editing and processing software and hardware. Even if there's only a little signal, if the drums were EQ'ed correctly and live in their own unique frequency range, what we would do at our studio is take the drum tracks, pull the different types of drums out as separate tracks if they weren't recorded that way, or were mixed down and discarded, and then boost the signal accordingly, clean it up, and remix the drums to create new tracks, which would then be mixed back with the guitars and bass, which I assume you kept separately.

That process is rather long, involved and requires a serious investment in gear and software, so it's beyond what I can get into here. You can try playing around with EQ'ing the drums tracks using the Tascam or running it back through the Yamaha console and hoping that the EQ for the bass levels is about the right range (helps if the console has a way to adjust that range, known as the "Q") and then record the playback again on the Tascam. That's a middle-ground solution that I would normally suggest, but when you say "I can hear a LITTLE bit of bass drum" then I think it's likely a lost cause. Again, sorry to say.

So sadly, you'll probably have to have post-production done on the drum track(s) or re-record them, and as said, either way there's a lesson learned, which is that you want to set things up so that there's as little gear between the mics on the instruments to whatever is getting the sounds to your ears. Remove as much as you can as you set up the recording levels and EQ before you start adding any processing, effects, or even a recording device like the Tascam 2488.

In short, if you want things to sound good, keep the recording chain as minimal as possible until you get the right sounds, then add pieces to it one by one, checking constantly that you're still getting the sound you want. If not, then either you need to fix or replace the gear you've added, or consider whether or not its needed or is necessary. So many problems that take a lot of time and money to fix in post-production can be avoided if just this little extra time is taken during the original recording process.


Posted by Wink Junior at 04:28 PM | Comments (0) | TrackBack

Yet More Selling Prices for Roland RE-Series Space / Chorus Echoes

Roland Space Echo RE-100 - $500 plus $120 (!?) shipping.

Roland Chorus Echo RE-301 - $600 plus unknown shipping.

Roland RE 101 SPACE ECHO - $375 plus $20 shipping.

We're quite confused by prices - the RE-100 is rare, indeed, but a lousy unit that is rare only because it wasn't very good and was quickly replaced by the RE-101. To see an RE-100 sell for $620 but an RE-101 sell for less than $400 leave us to just feel that Ebay isn't really the place to find out what things are worth. They vary widely. Like in the same week, watching one SRE-555 sell for almost $1300 and another one, in better condition, sell for $650.

Posted by Rob V. at 11:52 AM | Comments (0) | TrackBack

April 10, 2007

Good Books to Learn MIDI

The whole MIDI setup is very easy once you get over the hump of figuring out how it works - it's far less complicated than it seems. That said, here's a link to a list of books we recommend to people who want to learn MIDI:

MIDI Books on Amazon.com

Any one of the first 4-5 books should be enough to really get a good understanding of how MIDI works.

Posted by Wink Junior at 11:01 AM | Comments (0) | TrackBack

April 09, 2007

Q: How Do I Use MIDI with my Keyboard and Cubase?

Niki wrote:

"Basically, I pretty much already have everything I need: Cubase SX 3.0 program, MIDI In-out cable, MAudio soundcard. And I also already understand a bit of the basic stuffs. Now I want to record as MIDI (not WAV) from my Keyboard."

I apologize in advance if this isn't the best answer, but your question, while I appreciate the detail you put into it, is rather confusing.

First off, here's what I'm reading as what the problem is you're trying to solve: you want to play your keyboard, record the MIDI output in Cubase, and then play back the recorded MIDI in Cubase and have the keyboard play it, like a sound module. If that's not right, you'll have to post a follow-up.

The first problem I see is you say you have a MIDI in/out cable. You'll need two at least to do this. The MIDI out on the Korg needs to go to the MIDI in on your M-Audio soundcard, and vice-versa, the MIDI out on the soundcard needs to go to the MIDI in on the keyboard. So you need two cables to go back & forth with recording and playback.

Cubase will never show outboard gear as a VST, or know about outboard gear in *any* way. VST plug-ins are software and so Cubase can see and register them for use, but outboard MIDI gear is simply assigned a channel (1-16) or set to "OMNI" or "All Channels", which will send the MIDI to all instruments.

What you need to do is program the keyboard to be on a specific channel only, once you have the two cables set up, say "channel 1". Then you tell Cubase to record MIDI coming in from channel 1 only, assign a track to it, hit record, and play. Then on playback, you have the output from the recorded MIDI track go to MIDI channel 1, and since the Korg has been assigned that channel, if you have it set to receive MIDI (usually there's a "Local (MIDI) Only" mode you need to turn *off*) - it will receive the MIDI data and play it back.

So get your cables set up correctly, assign the keyboard a channel and set up Cubase to record and playback over that channel, ignore the VST stuff which has nothing to do with what you're trying to do, and you'll be set.

My guess is that you don't have two MIDI cables, and don't have the keyboard assigned a specific MIDI channel that you can record and playback in Cubase. Your Cubase manual will have all the details on how to do this, so just refer to that for all the settings for the program, once you have the MIDI cables and keyboard MIDI channel set up. Good luck!


Posted by Wink Junior at 01:27 PM | Comments (0) | TrackBack

Q: How Can I Stop My Guitar From Buzzing / Humming?

Someone wrote:


"I'm having grounding problems which are making it difficult for me to record a clean guitar sound through a Roland G20 guitar synth and a Boss GT6 effects unit. There's a bad buzzing sound that stops when I touch the guitar with both hands, at least one on the strings. Every time I lift a finger or try to change chords, there's a loud 'click' sound. I don't have this problem with my mics or other gear, just the guitar. I have a Line-6 Pod Pro which also has the same problem: the buzzing increases and decreases when I move around the room. Even if I do find a spot where the buzzing stops for a bit, the slightest thing will set it off again."

There can be several obvious things that can lead to a buzzing or humming sound from a guitar, especially if it's run through effects pedals:

  1. First and foremost, you don't mention if you have single-coil pickups, which are known for picking up any kind of electrical signal that might be nearby, although humbuckers are not completely immune from this either. If you have single-coil pickups on your guitar, you'll have to do your best to eliminate anything in your recording area that might be emitting electrical "noise" or signal.
  2. The power supplies to any effects are not being run off of a clean power source, and if you get a UPS or other type of "power conditioner" which will "clean" up the power supply, by making sure that the AC is exact 60 Hz (in the U.S., 50 Hz in the U.K.) and will even out any variations in wattage and block any spikes, which may also damage your gear. A UPS for every power source is pretty much mandatory in any home studio these days, since they're very affordable.
  3. One or more of the power supplies for your gear probably doesn't have a built-in power supply but instead a "wall wart" - an AC to DC transformer that's a box around the plug itself - these can often create a lot of noise/signal if they're poorly built. You can often figure out which of these, if any, are a problem by plugging each one in and moving the guitar closer and further from the power supply ("wall wart"). If it gets noisier as you get closer (but no closer than 1 ft. away) - then you should probably find a replacement for it, as it's not only noisy, it's probably supplying "dirty" power, despite the UPS it may be plugged into, to the device its for. Many of these supplies are cheaply and poorly made, and a good replacement is inexpensive and worthwhile.
  4. You don't mention if you have any CRT computer monitors near your gear, or if you have any external hard drives or other USB or FireWire computer devices. These all create a lot of electrical interference (one of our ext. hard drive backups we can hear spin up when we're mixing or processing audio or video.) In the case of the ext. drives, you can simply turn them off when you're recording, and in the case of CRTs, the best thing you can do is to replace them with a new flat screen monitor, which doesn't put out much noise at all, and uses a lot less power to boot.
  5. You may have one or more effects in the chain from the guitar to your recording unit that are noisy or bad. Some pedals are just noisy when turned to certain settings or to extreme levels, and some just go bad: Line-6 products are well-known for two things: being very great-sounding for a cheap price, and for eventually going bad, some times just dying, some times completely freaking out, and other times getting noisier and noisier as time passes. We'd definitely recommend checking out the Line-6 Pod Pro by taking it out of the chain and seeing if that clears up the humming.
  6. Even light dimmer switches, that let you adjust lights rather than just on/off can create a lot of interference, as well as Halogen lights, which are very noisy. You'll want to eliminate these or be fa r from them when you play.

If the hum changes in character and intensity as you move around the room, getting closer or further from computer equipment and your gear, or even when you face in different directions, this suggests that you are picking up radiated interference rather than something coming in via the mains — mains hum wouldn't be affected by your position in the room.

Incidentally, I've also come across some power adaptors that do put interference into the mains circuit when they're plugged in. So it could also be worth unplugging as much gear as you can, testing the guitar signal path, then, if it is quieter, plugging the other gear back in a piece at a time and see when the interference comes back. You may well be able to identify a faulty or sub-standard adaptor this way.

A probably cheaper option than replacing a lot of electronic gear and power supplies is to modify your guitar, installing humbucking pickups and lining the cavity which houses the pickups and electronics with insulating copper foil.

The other factor that may be aggravating the situation is the quality of the mains grounding, which is why a UPS or some kind of power supply regulator is crucial. It is worth using a three-pronged illuminated socket tester to check that you have ground in the first place, though that in itself doesn't guarantee good results. But at least you'll know you have a ground, and if you don't, then you'll have to start there.

As always, if you feel unsure about anything electrical, get hold of a qualified electrician!

Posted by Wink Junior at 01:27 PM | Comments (0) | TrackBack

April 08, 2007

New Gallery: Roland RE-100 & RE-101 Tape Echo Photos

We've started a gallery of photos of tape echoes we've owned and/or worked on, to share with the public. Comments will be added as we have time, and hopefully people can leave comments themselves to help share information about these machines:

Roland RE-100 & RE-101 Tape Echoes

Enjoy!

Posted by Rob V. at 07:03 PM | Comments (0) | TrackBack

April 07, 2007

Reverb After Mixdown - Why Post-Production Studios Can Help

Here's a recent response to a followup question on the AllExperts.com Home Recording forum, where we're moderators and "experts". We felt it was worth sharing as it shows one of the ways in which we provide post-production services that are really beyond what most home recording setups, as well as Recording Studios, which tend to dedicate all of their time and money into mics and recording systems like 2" tape or ProTools.

It also has what we consider some good advice on adding reverb to a track after it's already been mixed down, as well as how to record them in the first place and why too much reverb is almost always a Bad Thing (tm).

Someone named Alex wrote back, first to say he'd tried using the software reverb in Cubase first without luck:

"I've tried using cubase to add some reverb but it sounds pretty bad."

Well, if you want to buy an audio editing program that supports some kind of effects plugins, you have many options, most of which are not cheap. There's Adobe Audition, which used to be Cool Edit Pro, Sound Forge (Sony, was Sonic Foundry), Steinberg Cubase, and a few others, but just the programs are fairly expensive, and while both of those I'd recommend because they come with built-in effects that are of decent quality, some even pretty fantastic once you learn to tune them. But it sounds like you've got that route and not been happy with the results.

I'm a bit surprised - if you own Cubase, assuming it's not the LX "Lite" version, and heck, even if it is, and you're just adding some reverb then you should be able to get decent results. If I can say without offending, my guess would be that you need to study up on how reverb works, hardware vs. software as well, and how Cubase reverbs compare to other software models. My bet is you can get decent results if you spend enough time on it.

But then came the clincher:

"I have the tracks saved - I recorded them in a studio. But they have no effects whatsoever as I didn't have the chas for that. I was thinking I could import the audio files into a program that would allow me to add some reverb to it or something. But am In assuming I want to add an effect to the entire track? The vocals aren't recorded separately you see."

But you mention that you don't have the vocals recorded separately - so if you're just trying to add reverb to only the vocals, then you've got a major headache. We've run a post-production and mastering studio at The Sound-O-Mat and we use one of two different approaches:

0. We analyse the track to find out what freq. range about 95% of the vocals fall into and hope to heck there's no bleed-over from other instruments in the same ranges (guitar, saxophone, etc.)

1. The combo of Adobe Audition with Waves' Rverb or TrueVerb allows us to apply reverb to just certain frequency ranges, and then using the range above, apply reverb to that range, and try our best to keep the mix as it was.

2. If people are willing to pay the extra cost, which we think is worth it, we can extract about 95-99% of the vocals (and any bleed-over, unfortunately) and create two separate tracks, one w/vocals and one w/o, then we apply reverb and spend time adjusting the wet/dry ratio (the amount of vocals w/reverb vs. the amount of original vocals) and mix-down the tracks again. This often works surprisingly well - we've surprised many a client.

The biggest axiom when it comes to reverb is that "less is more" - usually, unless you're using it more as part of the instrument than not, you want it not to be noticeable. We recently met a producer whose "signature approach" turned out to be massive, heavy use of reverb on every single recording, regardless of what kind of music. To our ears, it sounded horrid - some instruments or vocals benefit from having the reverb noticeable, but in this case, it just dominated every single recording - it was rather annoying and poorly done, we thought, but that's why we're in post-production - to fix all the really Bad Ideas (tm) that Producers often have, or the lousy job that most recording studios that claim to do mastering do.

I'm not sure why you bring up mics, unless you plan on re-recording? If you are, then we'd suggest recording your tracks with at least one mono "completely dry" track, and then one with the reverb that sounds good at the time, and then mix those two, or add reverb to the dry track.

Other than that, it's quite a lot of work to try and do what you're suggesting - we have probably about 10 years and $4000 invested in software to do the above procedures, so all I can suggest is that you give it your best go, or consider trying to hire the job out to a post-production studio. Good luck!

Posted by Wink Junior at 10:56 PM | Comments (0) | TrackBack

April 05, 2007

April 01, 2007

Telling an old vs. new Roland RE-201 Space Echo unit

Roland Space / Chorus Tape Echo units were invented in 1978 (there's a great history of Roland which mentions this by Sound on Sound magazine) and were made until 1993. The RE-201 unit was by far the most popular and best-selling, although lacking the "sound-on-sound" feature of of the later models: the RE-301, RE-501, and SRE-555.

At some point in the mid-1980's, Roland made some minor and some important changes to the design, and you should take this into account when purchasing one: a latter-era RE-201's are better built and more reliable than the earlier ones; Roland was smart enough to improve all the weaknesses that they saw a lot of service problems.

The two major ways to quickly tell if you're dealing with an older RE-201 are:

1. The older versions do not have a three-prong plug with a ground. Without proper grounding, a lot of older RE-201's can develop a buzzing or humming noise due to bad grounding or grounding loops. So if you see just two prongs on the plug, you're dealing with the V1 ("version 1") model. Here's a photo of what a V1 model plug looks like:

[Roland RE-201 V1 Two-Prong Plug]

The newer V2 model has a standard three-prong plug, just like almost all modern electronic equipment.

2. The "Echo / Normal" switch on the front panel, that allows you turn turn the echo off, was thin with a white plastic cover, and broke often enough that Roland started supplying an extra one along with the extra tape loop. The latter-era V2 models have a much thicker, silver/chrome switch that's solid and is very unlikely to break.

Here's photos of the V1 vs. the V2 echo on/off switches:

Model V1:
[Roland RE-201 V1 Echo Switch]
[Roland RE-201 V1 Echo Switch]

Model V2:
[Roland RE-201 V2 Echo Switch]
[Roland RE-201 V2 Echo Switch]

There's also a number of important internal differences. The tape motor is different, the V2 one having a much higher torque rating, and tends to run at a much steadier speed - older models are more likely to develop "warble" or "wobble", which can also be caused by bad tape or the need for a good cleaning. The motors in the V1 version just aren't nearly as well-made.

Here's a list of other differences we've found:

1. The V1 model runs a lot hotter and you have to make sure it gets good air circulation. None of the tape echos have a fan to move air through the unit.

2. The V2 model transistors and caps (capacitors) are better and cleaner, and don't affect the sound as much (leaving a better analog and/or tape saturation sound.)

3. The foot-switch plug tends to work better - some times on the V1 model, you'll hit a foot switch and the unit won't turn on/off.

4. The output phono jack has an orange plastic circle around it on the V2 models, which just makes life easier in dark clubs. The input has a green plastic circle around it as well. Minor, but nice when you're in a hurry and staring at a row of phono-jack plugs in poor lighting.

Otherwise, they're more or less the same unit, but we feel a V2 is worth about $100 more than a V1 because the grounding plug really makes it less noisy and less likely to get fried if there's a power spike, because neither unit is well-fused, and a power spike can fry components on the circuit board of either unit. They did add fuses to the later models.

[Roland RE-201 Power Supply Fuse]

There were two things that we consider a step "backwards" on the V2 models vs. the V1:

1. The out-take caster on the V2 is smaller, lighter aluminum and doesn't hold the tape quite as steady as the heavier, metal ones on the V1.

[Roland RE-201 Out-Take Caster]

2. The V1 had clips to close the cover, vs. screws which you have to take in and out to open the unit. The change was made due to complaints of the clips opening up while moving the V1 unit, often spilling things inside, but we find if you keep the clips clean and adjusted, they're just fine, and the damned screws take too long to open and close the lid, and are easy to misplace and lose.

[Roland RE-201 Power Supply Fuse]

3. The V1 had a slightly bigger handle, and with huge paws like ours, we liked it better - both have metal inside the straps, but the V1 was longer and just easier to carry.

[Roland RE-201 Power Supply Fuse]

That said, our own RE-201 has a V1 caster and the clips on it, and a bigger handle from a V1, since we like them better, so our "hybrid" offers us the best of both models.


Posted by Wink Junior at 10:14 PM | Comments (0) | TrackBack