Instruments & Gear

Recording engineer Steve Hall explains why your old vinyl records really do sound better than some of what you're buying now.

This post first appeared in a longer form in the Disc Makers blog.

Since the earliest compressors were conceived and built, the ability to modify, control, and maximize the dynamic range of a musical performance has been the quest of many an audio engineer. In the early days of audio, limits on a recording’s overall dynamic range were dictated by vinyl – the state of the art in music delivery until the CD’s debut in 1982. Today, with virtually all music being recorded and massaged in a digital environment, it’s become standard operating procedure to limit, compress, or maximize the dynamic range of a track or an entire mix.

On the surface, having your track as loud or louder than the competition seems like a good thing in the world of car radio and MP3 devices. But in speaking with some top engineers from the music, broadcast, and mastering fields, maxing your levels is not a recipe for a great sounding recording. In fact, according to experts Steve Hall, Michael Bishop, and Patrick Fitzgerald, you may be doing more harm to your recordings than good through the liberal use of the compression and dynamic control programs found in every workstation across the country.

Steve Hall
Steve Hall’s Future Disc Systems mastering studio is in McMinnville, Oregon. Steve has mastered thousands of albums for artists such as Madonna, Chris Botti, Green Day, Sisquó, and the Grateful Dead, to name a few. Of these more than 100 have achieved gold or platinum status.

Would you say dynamics have been squashed out of music today?

They’re pretty much gone, yeah. So much of the music I hear today is totally hammered and the result is that most of the life, detail, and energy of the original performances are just gone.

How did we get to where we are today?

When vinyl was the dominant release format, artists and labels were always concerned about cutting their records as hot as possible. Factors that were limiting on vinyl were how much bottom end the disc could hold and how long a side could play. That said, most of the dynamics heard during the original recording session were pretty well preserved on the LPs. With 45s, you always wanted your record to pop when it came on the jukebox or radio, but not to the point that the performance was lost. So the artist, label, A&R guy, and mix and mastering engineers would all listen carefully and agree on some sort of compromise between maintaining the recording’s musicality and achieving the hottest level possible.

Today, with digital, there aren’t those kinds of format limits in place. There’s simply a brick wall looming at the end of the digital audio pipeline. With finalizer-type tools, and various multi-band compressors, the goal is to modify the master to constantly have everything as loud as possible, and you pay a price for this so-called “normalization.”

When did the shift occur?

You only have to go back about ten years and listen to the masters from that era, really anything before the mid-90s. Listening to those albums is a totally different experience. There’s life, energy, and detail in those releases that you seldom hear today. And the main reason is that now everything is just maxed out from the beginning of the album right to the final note.

Misuse of compression is even worse on television. Especially on cable TV, things are often abusively slammed.You are constantly reaching for the volume control on your remote to deal with a situation that is made worse by compression, rather than being improved.

Michael Bishop

Michael Bishop
Next up is multiple Grammy award-winning engineer Michael Bishop, of the renowned audiophile label Telarc Records. Telarc, based in Cleveland, OH, has won an amazing 40 Grammy awards for its outstanding recordings in classical music, jazz, and blues.

Michael, are dynamics dead in today’s popular recordings?

I’d say that in most non-classical recordings, the answer is yes. Load any CD into your workstation and look at the resulting waveform. You’re seeing rectangles where you used to see waveforms. There used to be dynamics there! And what’s amazing is that all the while, you look at some of the recording forums that go on, and every once in a while a thread will start up about old Frank Sinatra recordings or older rock recordings from the ‘60s and ‘70s, and these old recordings are being held in great reverence.

People are often referring to them as “the sound to go after,” all the while taking their own present-day recordings and maximizing them, hitting the “normalize” button, and cranking everything up to be maxed out all the time. If you go back to those classic recordings, compression was almost non existent. And yet they still sound loud and have tremendous depth. There’s an example where less is more.

Telarc is known for producing audiophile recordings. How do you go about recording and mastering to ensure your recordings have depth and life?

We do everything in high resolution, most using the Sony DSD [Direct Stream Digital] format. Our source masters are always DSD for all in-house engineered and produced sessions. We do have some outside-produced projects, and they are so accustomed to delivering a master with everything maxed out. So we ask them to go back and remove some of the compression because it doesn’t mesh with the rest of our releases, which tend to have a pretty wide dynamic range.

Going back a few levels of “undo” puts some life back into the recording. I don’t think it’s too much to expect the consumer to bring their volume control up a little bit. When a consumer puts a recording in a multi-disc CD player or an iPod with play lists that have everything thrown in together, levels from song to song become an important concern. In that context, if one artist’s release doesn’t hold up against another’s, that’s a problem.

How do you deal with that?

We’re releasing in two formats on many of our releases: the standard CD and the high-res SACD version, which will be in both stereo and surround. But we never produce square waves where there once was music.

We use a moderate amount of compression, listening carefully to the results as we go. We have two other engineers in-house besides myself, and we each approach things a bit differently. One of our engineers prefers to use the Sequoia system and the Waves software bundles that come with it. He often does the releases for our Heads Up label, which focuses on smooth jazz, so they will have a sound that is compatible with that genre.

On the other hand, I like to use the Sadie system and the TC 6000 for more of the straight ahead jazz, blues and classical releases that come out on Telarc. Regardless of the artist or release, we’re never going to get in a level war. About the worst thing we could do is to start comparing apples and oranges, instead we just make each release sound as good as it can for the specific format it will be listened to on.

You have to keep in mind that too much intrusion into the dynamic range of any music is taking out the life and taking out the fun of listening to it. It also makes it very fatiguing to listen to, no matter what the format. There’s a great fallacy out there that if you make the CD really loud, it’s going to come across much better on radio. It’s actually exactly the opposite.

Any time you listen to a jazz station, and you listen to a modern recording that’s been pretty maxed out and then you have a classic track come on – say Louis Armstrong or Ella Fitzgerald – that track just jumps out of the radio in comparison. That’s because it was not squashed when it was mastered and released, which once again demonstrates that less is more when it comes to using compression.

Patrick Fitzgerald

Patrick Fitzgerald
Over the past 20 years, Patrick has mixed thousands of short and long-form soundtracks including the famous MTV animated station IDs, hit video games, a number of feature films, DVD releases, and many thousands of commercials.

From your perspective mixing for broadcast or film, are dynamics dead?

I wouldn’t say they’re dead, but it is a daily struggle to make sure that there are some. Working with music is becoming more of a struggle, because what I am now getting is so compressed to start with. You look at the waveforms of many of the music tracks that are sent to me and everything is tucked right up next to digital zero and all the peaks are just cropped right off!

It’s ironic when you look back at the history of recording, how the value of dynamics in music has changed. Consider the evolution in the middle of the 20th century, from really poor quality tape recording to really good tape recording then to vinyl. During those years, there was always this quest to answer the question: How do we get more dynamics into the recording process? And now that we have a system where you can have absolutely incredible dynamic range, everybody throws it away.

Do you use compression?

I use compression all the time, but I’m careful to never let the compressor mix for me. I probably spend the most work evening out dialogue. Twenty years ago we worked primarily with professional actors and voice artists, people who spent years training to project and control their voices. They knew about breath support. You can hear anyone who’s had that training right away, because their levels come out much more even. A lot of the dialogue or voice over we get today is voiced by people that don’t have that training, so there’s a lot to do in order to make it all intelligible.

This post, by Keith Hatschek, first appeared in the Disc Makers blog. Hatschek has been a musician, educator, recording engineer, producer and marketing executive. He is the author of  The Golden Moment: Recording Secrets from the Pros and How to Get a Job in the Music Industry. Hatschek currently teaches Music Management at the University of the Pacific.

Stephen Wise is your guide to home recording.

In this post, we will begin to explore the Audacity® Effects. If you’ve never used one of the “pro” tools, then you are probably better off since the limitations of Audacity® will be less apparent. In Audacity®, the graphical interfaces to the plug-ins are quite primitive.

Primitive interfaces, coupled with a lack of a “real time” option makes set-up a tedious trial and error affair. That said, since I also develop applications, I really do appreciate what the authors of Audacity® have done. They’ve foregone the slick Graphical User Interfaces (GUI) for plug-ins in favor of functionality.

When you’re developing as a hobby without the benefit of a well-paid full-time development team, things just have to happen more slowly. I might have done a few things differently for myself, but it’s obvious that Audacity® is aimed at an audience who has limited experience. It’s all free, of course, so you can’t say that you were cheated! If you’re working for money, though, professional tools will pay for themselves over and again in time saved.

One of the workhorse modules of the traditional recording studio was the parametric equalizer. Most people are familiar with the graphic equalizer, where each control amplifies or attenuates a specific band of frequencies, and with tone controls, which are shelving equalizers. The parametric equalizer is a “dial up what you want” filter. One control selects a particular frequency at which filtering will begin.

Another control selects attenuation, preferably separate controls for input and output. A third control sets “Q”, which is how sharply the effect cuts in. An inversion switch completes the basic controls. Using the controls, one can, at any selected frequency, amplify or attenuate lower frequencies, amplify or attenuate higher frequencies, and amplify or attenuate a band of frequencies. Audacity® doesn’t actually come with a parametric equalizer, but it has two filters in the basic effects that can simulate parametric equalizers pretty well. Since you can add your own plug-ins to Audacity®, you can have an actual parametric equalizer if you want one.

If you drop down the Effects menu in Audacity® (remember that you must select some or all of a track first), you will see two filters in the basic effects: “Equalization…” and “FFT Filter…”. Both of them work pretty much the same way, so I will limit this discussion to the more comprehensive Equalization. When you open Equalization, you will see at the top a window with a horizontal blue line. To the left are markings in decibels (dB). The decibel scale is a log scale as opposed to a linear scale. A level of 24dB is quite a lot when you consider that you roughly double (or halve) the level for each 6dB! The dB scale is quite convenient for audio since audio covers a wide dynamic range. Your graphics will be much more meaningful in the dB scale. You can change your tracks to dB by dropping down the menu to the left of the track and selecting “Waveform (dB)”.

In the Equalization window, if you click along the blue line at the frequencies of interest, you will create deflection points on the line. You may then use the mouse to drag sections up or down. Try it! Use “Preview” to hear what you have created. You will probably notice immediately that when you drag upwards, you get a lot of distortion. Remember, 24dB is a factor of about sixteen to one! If you’re going to boost, be sure to first attenuate your track. As a practical matter, you will get better results attenuating than boosting. Boosting also amplifies the noises in your track. Attenuate whenever you can.

You have also noticed that Equalization has some presets. These presets are evidently to be used in converting old recordings to digital. Select a preset and Click “Load Predefined Curve.” Since RIAA has been used for a long time, it’s probably already built into the preamp that you use with your turntable. You really can’t make use of these presets unless your phono preamp is “flat” (i.e. unequalized).

The best way to get familiar with filter effects is to play with them. Whereas “Amplify” affects everything, filters affect only a band of frequencies. It’s like having tone controls that do what you really want. “Bass” controls on most equipment boost all of the way up through mid and even high bass. With these filters, you can just bring up the bottom end, or, bring it down if your smaller speakers can’t handle it. Many singers “hide” behind the band.

Use the filter to boost in the 1000Hz to 3000Hz range to flush them out. Do you have a low “hum” or “rumble” in the track? Find it and pull it down. Hum and TV buzz are usually at 60Hz in the USA, but often at 120Hz. Rumble can be removed by pulling the very left end of the blue line all of the way down. Likewise, pull the right end all of the way down to get rid of “hiss.” There’s an interesting tool for finding frequencies to filter. I will discuss it next time.

Stephen Wise has been designing electronic musical instruments since 1975. Steve specializes in realistic recreations of traditional instruments. He became interested in the field after hearing Walter/Wendy Carlos’ “Switched On Bach” and upon being introduced to the Allen Digital Computer Organ, the world’s first all digital musical instrument. Steve is currently designing instruments for Schulmerich Bells, maker of handbells, electronic carillons, and the breakout MelodyWave® instrument.

"Gear Guy Chris" aka Chris Grova is a veteran collector of instruments, gear, and gizmos. He shares his knowledge about what to buy, or what to dream about buying.

No matter what you call it, ‘whammy bar,’ ‘trem bar,’ or whatever, there’s only one (IMHO anyway) that’s stood the test of time for over half a century and remained the arbiter of cool: the Bigsby. Exuding an old-school vibe that still resonates today, the Bigsby Vibrato adds classic functionality to many an electric guitar. 
 
The vibe
The device itself is a study in artful simplicity: a spring-loaded arm attached to a pivoting metal bar where the strings attach, mounted on top of the guitar. Press down, the strings loosen and the pitch lowers, pull up to sharpen the pitch. The Bigsby is best suited for tasteful, subtle downbends rather than extreme ‘dive bombing’ ala the Floyd Rose and other locking system.
 
The Bigsby vibrato comes factory-installed on select guitar models from manufactures such as Gibson, Fender, PRS, and Rickenbacker. However it’s with Gretsch that the Bigsby is most often associated (over 55 years).  It’s not surprising that after decades of collaboration, when the opportunity presented itself, Gretsch purchased Bigsby in 1999.
 
And while we’re at it, let’s clear up a common misconception: vibrato vs. tremolo. Unlike tremolo, which is a fluctuation of volume, vibrato is the fluctuation of pitch. So, while you may have heard the generic ‘tremolo bar’ term thrown around – the effect produced is actually vibrato. And, while the Fender Vibrolux is a great amp, the knobs directly to the right of reverb knob  actually control  the tremolo effect – not ‘Vibrato’ as the name would suggest (but no need to split hairs….we get it.)
 
Some history
When they build the Mt. Rushmore honoring 20th century American musical pioneers, Paul A. Bigsby will be right up there with Leo Fender and Les Paul.  Bigsby is indeed a founding father. While Les Paul’s ground breaking “The Log” was one of the world’s first solid body electrics, it remained more of a prototype until the Gibson company eventually warmed to the concept. 

That warming was no doubt fueled by watching Leo Fender’s success in mass marketing its Telecaster and Esquire models. However, before Fender got their arms around the solid body electric concept, it was Paul Bigsby, initially a builder of steel guitars for country-western swing players of the day, who built what’s considered the first modern solidbody electric. 

The date was May 25, 1948, and the guitar was for country musician and guitar legend Merle Travis. A few years later at Travis’s request, and to improve upon an existing device, Bigsby designed and built the mechanism that we know today as the Bigsby True Vibrato. The ‘True’ bit means it’s possible to alter the tone above and below the normal pitch.  Today, the Bigsby remains a classic – some might say de facto — choice for players and guitar manufactures worldwide.
 
According to the Bigsby folks, “For any guitar that matters, there’s a Bigsby to top it off.” Here’s a general guide courtesy of Bigsby Guitar:
• Model B-3 is designed for thin electric guitars
• Model B-5 is designed for flat top solid-body guitars
• Model B-6 is designed for large acoustic and arch top guitars
• Model B-7 is designed for thin electric guitars with more downward pressure
• Model B-11 is for thin electric guitars with an arch
• Model B-12 has a pressure bar and is designed for large acoustic and arch top guitars

Players like Chet Atkins, Brian Setzer, Neil Young, Johnny A., and scores of others have incorporated the Bigsby sound as a signature. Perhaps not for everyone, but for some of us, a Bigsby tailpiece helps define an individual sound, style, and approach. 

To quote the late great Les Paul,“When I put the first Les Paul out I put the Bigsby on there. I have it put on all my guitars, it just happens automatically. The Bigsby just goes with the Les Paul guitar!” I took Les’s advice and put a B7 on my ‘79 Les Paul Standard — your humble scribe couldn’t be happier.
 
For more info, visit Bigsby Guitar or check out the fantastic book: The Story Of Paul Bigsby, Father of the Modern Electric Solidbody Guitar by Andy Babiuk.

“Gear Guy Chris,” aka Chris Grova has been providing shelter, love, and a good home to wayward guitars, amps, effects pedals, and other assorted musical gizmos for over 30 years. Luckily, his wife and neighbors don’t seem to mind.

The old ribbon microphone is making a comeback; modern versions with classic designs can be purchased today.

This post first appeared in the Disc Makers blog in a longer form.

There are some astounding values placed on vintage instruments and recording equipment these days. A 1958 Stratocaster in excellent condition, for example, may fetch as much as $25,000. Vintage recording devices from bygone years may also be valued at $10,000 or more for the most coveted items, such as rare German tube mics or broadcast limiters.

For the vast majority of people, these prices put items like this out of consideration. But researching, locating, testing, and purchasing lesser-known instruments and devices from the past can yield some fantastic results.

In our search to learn about some of the less-obvious paths that a person interested in such old gear might travel, we spoke with Mark Rubel, owner of Pogo Studios, located in Champaign, Illinois. Mark has been finding and collecting vintage instruments and recording equipment for many years, and his studio is home to dozens of rare and wonderful items. His recording credits include projects with Hum, Alison Krauss, Adrian Belew, Melanie, Luther Allison, and Henry Butler. He also teaches music technology and recording at Eastern Illinois University, where his students often ask him about his interests in vintage gear.

Classic sounds
“There are certain pieces from what we might call the golden age of technology, the mid-1940s to the mid-1970s, that have become classics,” says Mark. “Why are they classics? Maybe they offer a particular sound, one that is immediately recognizable, or are ergonomically designed and are intuitive to use. Beyond those reasons, I like vintage gear because of the cultural, emotional, and psychological value they may offer. I think there’s a visceral reaction, a kind of immediate feeling you get when you hear some of these pieces of gear. For instance, a vintage Tele, a Vox AC-30, or a 60s combo organ. You hear it and say ‘There it is! That’s the sound!’

Ribbon mics are back
In our discussions about vintage gear in general terms, Mark identified some rare gems and a few affordable pieces. “Many of the older microphones are iconic and highly prized. All of the old ribbon microphones from RCA are classics, along with a few of their dynamics. Old tube mics made by Neumann, AKG, and Telefunken are all collector’s items today.

“I’m pleased with the comeback of ribbon mics, today you can buy very nice ribbon mics from Coles, AEA, or the Cascade Fat Heads, which offer great value. I’d even call them modern-day classics. I think the comeback is due in part to the fact that ribbons are well-suited to the type of recording medium we now use – digital, they have a naturally dark sound, in part because their design. Condenser mics that were the gold standard in previous years, were better suited to recording on analog tape. The condensers have a resonance around 10-12K that helps minimize the high end loss that is inherent in analog recording when you are dragging the tape across the heads hundreds of times making a record.

UREI Compressor

Compressors in demand
“The UREI compressors still sound amazing today. I was just using my LA-3A limiters the other day and marveling at how much you can compress a signal and still have it sound very musical. Any of the old tube broadcast compressors made by Collins, Gates, or Fairchild are rare and in great demand. Old spring reverbs are fun and not very much money.

I’ve grown fond of the Roland Space Echo. It has a nice sound of its own. I always make sure to show my students how it works and to also route it back into itself to show them how tape feedback works, speeding it up and slowing it down – which is basically the sound from nearly every sci-fi movie from the 1950s.”

Guitars made simple
 “When it comes to old guitars, beyond the Gibsons, Gretschs, and Strats which are so expensive, I’m fond of Baldwin guitars. They were made between 1965-1970 by Burns of London, who made their own models as well as the Vox line of guitars. The guitar player in my band has a solid body Baldwin electric with two or three pickups and a toggle switch labeled, ‘Jazz – Rock 1 – Rock 2 – Wild Dog’ – that’s pretty cool!“

Rickenbacker Model 331, at left, and Ovation Magnum bass.

“At one point in the 1970s Rickenbacker made an electric guitar – Model 331 – with a color organ inside, so as you played, the guitar created its own light show. A few Japanese guitars from the 1970s are interesting, especially those by Tiesco and some of the early Yamahas, which were space-age looking. Ovation also made some solid body guitars with ridiculous names that were really ugly but well made with excellent wood and a lot of mass. I have a Magnum bass from that line and it has a built-in graphic equalizer, a graphite neck and a big rubber mute to get the deadened bass sound popular at that time. Two companies that are making affordable retro-style guitars are Eastwood and Italia, both of which have that cool 1950s look, an homage to some of the classic solid bodies from that era.”

Garnet Herzog

Vintage amps
“Old amps come in various sizes, but if it has tubes, it’s likely to be worth trying out. Two lesser-known brands are Magnatone and Garnet. Garnets were made in Canada. I think that Steve Albini and I may be the only people who collect Garnet amps. (Ed. Note: Thomas “Gar” Gillies, a sound technician who collaborated with the Guess Who and Bachman-Turner Overdrive, made them.) There was the “Pro” model, and the BTO (Big Time Operator), which had really interesting labels for its tone controls. They were labeled “Sound Fountain,” and the distortion control was called “Stinger.” They sound a bit like vintage Orange or Hi Watt amps from that period. You can find them from time to time on Canadian eBay.

“I’ve got a Garnet Herzog, which is a small tube amp that you can use with a separate cabinet. It’s about 15 watts. However, it was really designed as a pre amp to get loads of sustain by hooking its output up to another larger tube amp’s input. In fact, the classic sustaining solo guitar featured on the Guess Who hit “American Woman” was cut with Randy Bachman’s Herzog.

Retro synths
“Older synths can be fun to play with – I have a vintage Moog 900 five-oscillator system and an ARP 2600 in my collection. I also love the sound of an old combo organ, like the Vox Continental. There’s a certain obnoxious whine that comes from it that screams 1960s. Stomp boxes can be fun to collect, as they are small and relatively affordable. I have an Electro-Harmonix guitar synthesizer, which has an octave pedal and a fuzz, but a few years back, they reissued it, so mine isn’t as rare anymore. It’s has a triggerable hi- or low-pass filter which is pretty cool. I had a Foxx Tone fuzz for a while which was nice, but I ended up giving it to Adrian Belew who uses it now.

“Actually, today there are a lot of nice boutique pedals being made by small shops, so there are both new and vintage pedals that are collectible. The Danelectro line of pedals have some very good products in the $30 to $50 range like the Chili Dog and Tuna Melt, which are fun to use. I do think the weird parts that some of the older pedal manufacturers used have aged and now sound even weirder. For instance, if you have an original Cry Baby wah, its squally-sounding and can really honk more than a new one.”

Shopping tips
So is there any overriding philosophy Mark would offer for collecting and using vintage gear?

“Just explore and see what’s out there. Use your ears and if it sounds good or weird, it maybe worth purchasing. Basically, I would say if it has any of these three things, a big knob, some type of meter, or a tube in it that lights up, it’s probably worth buying if it’s less than $50. If it’s a pedal, or wah-wah, or some other small device and it works, I’d say $20 is a good range. Craigslist has become the new garage sale and has really replaced eBay as the place for finding these things.”

The final question that often comes up in discussions about vintage gear, is whether or not it really is better than the new gear being made today. Mark says: “Very likely, yes. The main reason is that in the case of what we would use for recording, it was built to either a broadcast or a military specification that was intended to last for 50 or more years. My friend, engineer Bob Ohlsson, reminded me that companies like RCA, Gates, and Collins actually leased their equipment to radio stations, so they really overbuilt their gear, because if there was a problem, they would have to go out to fix it. As a result, it was really designed and made with the best possible quality components.

“Basically, if it’s got a tube in it, it likely will sound better. Tubes do have a pleasing effect, but to work properly, pairing them with a good quality transformer is essential.”

This post, by Keith Hatschek, first appeared in the Disc Makers blog. Hatschek has been a musician, educator, recording engineer, producer and marketing executive. He is the author of  The Golden Moment: Recording Secrets from the Pros and How to Get a Job in the Music Industry. Hatschek currently teaches Music Management at the University of the Pacific.

Stephen Wise is your guide to home recording.

When exchanging, saving, or archiving files, file formats become an issue. Not only is there a trade-off between file size and quality, but some devices may not be able to recognize some formats. Computers have the advantage in that you can easily install new software to handle just about any file format known to man, but doing so may be an inconvenience and may be beyond the capabilities of some users.

Portable devices can sometimes be upgraded, but upgrades may be difficult or dangerous. The most common formats, wave and mp3, might seem to be universal, but they’re not. Whereas just about any device can handle wave files sampled at 44.1kHz 16 Bit, many will not handle “pro” formats such as 48Khz 24 Bit or 96kHz 32Bit Floating Point.

If you want to make your files as universal as possible, then you may want to stick with 44.1kHZ 16 Bit wave files and 64kbps, Fixed Rate mp3 files. If you create a file for your mp3 player and it doesn’t “play,” then looking into the format is a good place to start.

An observation that I have made is that sometimes a file exported in a “pro” format actually sounds worse than a file exported in a lesser format. If it happens to you, then it might be a codec issue. The codec in your playback device may be hard-wired for just one “native” format. When you throw it a file in some other format, the file is first converted to the “native” format, then played back. If the conversion algorithm in the playback device is of poor quality, then the audio suffers.

Many cheap or “on-board” sound cards actually only play back wave files at 44.1kHz 16 Bit. If you try to play anything else, then the file is first converted to that format. There are similar issues with mp3 files. For mp3, a variable bit rate can get you a smaller file at the same quality or a better quality for the same file size. Many devices do not handle variable bit rate mp3 very well. The solution to all of these situations is to discover the native format for your playback device and to export all of your files in this format.

If you’ve been playing with Audacity®, then you may have seen the export option “Ogg-Vorbis.” What the heck is “Ogg-Vorbis?” If you’ve been reading my blogs (see previous posts on right-hand side under ‘Instruments & Gear Recent Posts’), then you know that the license to use “mp3″ commercially must be granted by Thomson. Ogg-Vorbis is an open source alternative that you may use any way that you wish, free of charge. To my ears, it’s superior to mp3 for the same file size. Ogg-vorbis used to be for the kind of people who use Linux, but it’s becoming main-stream. It’s a great way to side-step the licensing issue. Certainly, any PC or MAC can be equipped with Ogg-Vorbis capability, and many mp3 players (non iPod) can handle it now (check the specifications).

Some of you have expressed an interest in remote collaboration by sending tracks via email or file sharing. File exchange presents a difficult trade-off. Track files are large. If you convert the files to mp3 or Ogg-Vorbis before sending them, then quality deteriorates with each turn-around and your final result will be less than satisfactory. Many audiophiles are now using FLAC, a lossless, open source file compression algorithm developed especially for music.

Since FLAC encoding is completely lossless, it is 100% reversible. Since FLAC does not discard anything, FLAC cannot achieve the same level of compression that can be achieved by mp3 or Ogg-Vorbis. The amount of file size reduction will be more on the order of what you can achieve with “zip” on a text file. Some audio files are more “FLACable” than others. Classical, acoustic jazz, and individual instrument tracks are highly FLACable. “Wall of Sound” or metal will compress very little. You will likely cover a range of 10% to 70% file size reduction. Any size reduction is worthwhile on a large file as file size greatly impacts total download time and gets you below size limits on email and free file sharing sites.

You can obtain the FLAC algorithm at FLAC download. Unless you enjoy using a “console” interface (typing in command lines), you might want to download a “GUI” such as “FLAC Front End.” Let me know what results you get with various content. I haven’t tried everything.

Finally, when recording or archiving material, you want to keep the quality as high as possible. Storage is becoming less expensive and computers are becoming more powerful. In the future, you will be happy that you saved high quality, unprocessed material. Even the processing that you do within Audacity® is irreversible outside of an Audacity® Project. Although it’s tempting to archive in FLAC, be aware that any compressed format is less robust than an uncompressed one. A lost byte in a FLAC file will blow a bigger hole than a lost byte in a wave file. If you’re following best practices and are backing up all irreplaceable or difficult to recreate files, then FLAC should be OK.

Coming up: We will explore some of the items in the Audacity® Effects menu and how they can help you fix problems and create more professional recordings.

Stephen Wise has been designing electronic musical instruments since 1975. Steve specializes in realistic recreations of traditional instruments. He became interested in the field after hearing Walter/Wendy Carlos’ “Switched On Bach” and upon being introduced to the Allen Digital Computer Organ, the world’s first all digital musical instrument. Steve is currently designing instruments for Schulmerich Bells, maker of handbells, electronic carillons, and the breakout MelodyWave® instrument.

Moving the mic 12 inches from the sound hole provided a balanced tone.

This post first appeared in the Disc Makers blog in a longer form.

If you’re doing home recording, one of the main instruments you may be using for accompaniment is the acoustic guitar. Learning the basics of acoustic guitar recording requires that you find your instrument’s sweet spots for micing, and also understand the recording environment.

We’ll use the most popular dynamic mics that many musicians rely on for gigs, the venerable Shure SM-57 and 58, to show how to get a good recorded sound from your acoustic guitar. We’ll also recommend two affordable condenser mics that can help you take your guitar’s sound to the next level.

Speaking in a 2004 interview, Grammy-winning engineer Richard Dodd, whose credits include a wide range of acoustic guitar-playing artists such as the Dixie Chicks, Tom Petty and the Heartbreakers, and The Traveling Wilburys, said: “Most great rock records have an acoustic guitar even though you may not notice it.” So how do you go about getting your guitar sounding good when you’re making the recording?

Think strings
First, be sure to put a fresh pair of strings on the instrument before your recording date. Oxidation occurs normally and deadens the sound of your strings, to the point where your guitar isn’t able to produce the full sound of which it’s capable. Personally, I’m partial to the Nanoweb-coated strings put out by Elixir, because they maintain their new tone longer and seem to produce a bit less finger noise which can become distracting when recording.

Next, be sure you are comfortable with your performance location before starting to record. I prefer to record standing up, using a strap, as I feel my posture is more balanced than sitting on a stool.

If you’re recording at home, try your living room area, because it offers a mix of absorptive (carpet, furniture, curtains, etc.) and reflective surfaces (walls, windows, ceilings, etc.). I have a reliable tuner handy to quickly confirm intonation between every take, to avoid nailing the perfect performance only to find that my “B” string had gone flat.

Before you get out your microphone and start to experiment, try the technique that many old-school engineers use when sizing up a new acoustic instrument and selecting the best mic and placement. They have someone play the guitar in the studio while covering one ear up and using the other ear like a microphone, moving their uncovered ear around the surfaces of the guitar at a distance of about 18 inches.

As you do this, you’ll hear the overall tone and frequency response of the guitar vary quite a bit. Move closer in and farther back to find the spot that offers a good combination of warmth and richness in the lower range, while maintaining some of the sparkle and shimmer that your freshly strung acoustic guitar has. When you find this “sweet spot” note the distance from the guitar body and then get your mic placed as close to that sweet spot as possible.

One final, but very important consideration is to ask yourself just what kind of an acoustic guitar sound will work best on the recording you envision. For instance, a demo featuring a solo voice and guitar would need a full-bodied sound to create a solid base for the vocalist, whereas a dense and heavily produced pop track may only have sonic space for a thinner, jangly-sounding acoustic that provides a little texture amongst the many other instruments making up that recording’s sonic landscape.

In the studio
Engineer Jeff Crawford started our test session with the “one-eared listen.” I was playing Jeff’s Takamine FS360 dreadnought guitar while Jeff evaluated its sound and decided which spots he wanted to mic. We had decided to use two of the most common mics found in any musician’s gig bag, the ubiquitous Shure SM-57 and SM-58, which would probably not be the first choice if you were recording in a top professional studio (most pro studios record acoustics with more expensive condenser mics), but because almost every musician has one or both of the Shure workhorses, it seemed sensible to see just how good a sound we could achieve with them.

Over the sound hole
Jeff started out with the SM-57, known primarily as the go-to mic for snare, toms, guitar and bass amp micing. He placed the mic over the sound hole; about 4 inches away and we proceeded to record a rhythm guitar part. The sound was full and present, but had a very noticeable increase in the lower frequency range, due to what’s known as the proximity effect, which every dynamic mic has to a greater or lesser degree. The proximity effect boosts sensitivity to bass frequencies more and more the closer the sound source gets to the mic capsule.

Twelve inches away
Next, Jeff moved the mic back about 12 inches from the sound hole. Playing the same part, we got an evenly balanced sound that had plenty of bass, but now had the guitar’s silky mid- and upper-frequencies in equal measure. “It’s pretty much what the guitar itself sounds like in the studio,” said Jeff.

Four inches away
We then placed the mic behind the bridge and about four inches from the soundboard, an area known as the lower bout, favoring the treble strings just a bit. This spot provided only the mid- and upper-midrange frequencies, with a noticeable absence of the higher overtones and missing this guitar’s rich lower range.

Six inches away
Finally, we put the 57 up near the spot where the neck and body joined, about 6 inches away and angled slightly toward the guitar body. As expected, we got a much brighter sound that emphasized the upper mid-range of the instrument, but we also picked up a slight boominess from the body. With a little low end EQ cut, this spot might be suitable for that rock track that already has plenty of other instruments in the low end, and that would benefit from a crisp-sounding acoustic filling in between the backbeat.

Next, we switched to the SM-58, a mic designed for live vocals, but occasionally used on other instruments. For some musicians just getting started with home recording, the 58 may be the only mic they own. Compared to the frequency response of the SM-57, the SM-58 doesn’t have quite as much boost in the upper mid-range, since its primary use is as a stage vocal mic.

We decided to use the exact same placements as we had with the 57, and found out that due to the mic’s differing frequency characteristics, the contrasts were noticeable. Starting out just 4 inches in front of the sound hole, we got a sound that was bland and without much character. It was dominated by the guitar’s low end, once again due to the proximity effect. However, there was even less of the mids and highs than with the 57 due to the 58’s built-in windscreen, and its different response curve.

Moving out to the 12-inch spot, we got a more even tonal response, but it was a little dull sounding compared to the 57. Our third test spot, on the lower bout, favoring the guitar’s treble strings, surprised us as it was a pretty complete guitar sound, not as bright as the 57, but definitely usable. Finally, we moved up to approximately 6 inches away from the neck-body joint and found that in this spot, the 58 delivered a pretty good image of the Takamine, but due to the closeness to the neck, we were hearing a lot of string fingering mechanics. The tonal balance was OK, so moving the mic a foot or so away would likely retain the tonal quality but minimize the mechanical sounds to a degree. For a quick solution on the SM-58 the lower bout offered the best bet, followed by some variation in distance and placement on the neck-body joint.

One other option you may test is to try recording using a combination of the mic and your guitar’s internal pickup if it has one. Many acoustics come with a built-in pickup to help with amplification in live settings, and these pickups, depending on their sensitivity, can sometimes enhance a guitar recording. Your internal pickup will usually give its best response when all the tone controls are set flat and the volume is set full on. Pickups tend to deliver a bright sound, so if you’ll also use a mic, you might want to try experimenting with the lower bout position mic and then mix the two sounds sources to get a proper blend between low and high frequencies.

After our test recordings, we got together in the control room to A/B the results of all four positions for each mic. All three of us agreed that for the 57, a foot away and directly in front of the sound hole delivered a clean, rich tone that would be perfect in almost any home recording. For the 58, we got the best results up at the neck-body joint, again, about a foot away. Remember to take the time to test out various positions with your particular guitar, because each instrument will have a slightly different “sweet spot.”

Condenser mics
As we started packing up, the conversation came back to studio condenser microphones and Jeff offered a few suggestions to consider when the time is right for you to add a higher-quality recording microphone to your home studio, so long as you have the necessary phantom power.

“I’ve had very good results with the A-T 4041 on many acoustic instruments. It’s a small diaphragm cardioid condenser mic that also works very well for drum overheads. Another reasonably priced condenser mic is the Studio Projects C-1. It has a large diaphragm, so it’s a bit warmer sounding than the 4010, which is very nice on most vocals and also good for acoustic guitar, and piano. The Studio Projects C-1 is an affordable large diaphragm condenser that can be used on a wide variety of sound sources in the studio. It will work nicely with any instrument where you are looking for a crisp sound with accurate low end.” (The street price on the 4041 is $299, while the C-1 can be found for $199, including a shock mount.)

This post, by Keith Hatschek, first appeared in the Disc Makers blog. Hatschek has been a musician, educator, recording engineer, producer and marketing executive. He is the author of  The Golden Moment: Recording Secrets from the Pros and How to Get a Job in the Music Industry. Hatschek currently teaches Music Management at the University of the Pacific.

Stephen Wise is your guide to home recording.

Mixing in Audacity® is easy. You create or import as many mono or stereo tracks as you need, move them, set their levels, pan (if stereo) and export to create a file that anyone can listen to.

When mixing, you will be interested in the tools at the left of each track as well as the Silence Selection Tool, the Trim Outside Selection Tool, the Time Shift Tool, and some plug-ins called “Fade In” and “Fade Out.” For any of the tools, if you hold the mouse pointer over them, a description will appear after a few seconds. To start, import any audio track that is on your PC (everyone has some) using the drop-down menu “Project” and “Import Audio…” . I will use some tracks from Incompetech. 

The box at the left of a track has some controls and information. There is the “X” for “close track window”, a drop-down menu, basic information about the track, and some controls. The “Mute” button silences the track during playback. The “Solo” button silences all tracks except the track whose Solo button you pressed. The horizontal slider marked -____+ changes the Playback/Mix level without changing the data in the track. The slider marked L____R move the sound towards the left or right channel during Playback/Mix without changing the track data.

A common task is to shift tracks in time. The “Time Shift Tool” (looks like a double-ended arrow), when active, will move a track forward or backward in time. You will use this tool frequently to line up tracks and to make tracks start after a certain time delay. Use the mouse to place the Time Shift symbol over the track of interest and, while holding down the left mouse button, drag the track left or right (see Figure 1 at right). If you require precise alignment, then you should first magnify using the Zoom Tool.

Professional mixes usually feature “Fade In” or “Fade Out”, even if the sound seems to start and stop instantaneously. Fade is also useful to bring a sound up or down from total silence without it sounding as if you just flipped a switch. There are multiple ways to do fades in Audacity®, so I will show you my favorites and you can explore others on your own. I normally use the “automated” fade that is in the “Effects” drop-down menu. These are called “Fade In” and “Fade Out” and are very simple to use. Simply drag the mouse over the part of the track that you want to fade (highlighting the area), then select either option from the menu. To remove sound before or after a fade, just highlight the area to silence and click the “Silence Selection Tool.”

The Fade effects will likely handle all of your needs for “zero to max to zero” fading, but there may be times where you don’t want to fade all of the way or you want to change to different levels at different times. The “Envelope Tool” (looks like two triangles pointing up and down with a line through them) is perfect for this task! You will have to play with this tool a little before you get a feel for it. When you activate the Envelope Tool, You will see blue horizontal lines at the top and bottom of each track. Clicking the mouse in a track will mark a “deflection point”. You can mark as many of these points as you wish. Moving to one of the points and dragging up and down with the mouse will “fade” the sound. It’s hard to explain in words, but it will quickly become obvious once you try it (See Figure 2 at left). This tool is great for bringing instruments or voice up and down. It’s also useful in creating samples.

Sometimes, you will want to change the level of the entire track or a large section of it. For this task, you will want to use the “Amplify Tool”, found under the “Effects” menu (Figure 3). If you do not allow “clipping”, then you cannot amplify beyond the point that the greatest peak reaches maximum. There is no case that you would want to clip. It’s just ugly! If you need higher levels than possible without clipping, then you will have to amplify using a compressor, a process that will be discussed in a future blog. Note that if you “Amplify” only part of a track, there will be an abrupt change in level.

With the tools that we have discussed so far, you can create a wide range of interesting audio files. In future blogs, I will get into the tools that handle special problems and, when used tastefully, can produce a professional sounding product. I will also peruse the Music After 50 Forum with the aim of addressing specific issues that you may have. Note from LRG: If you haven’t yet registered, it takes just a minute; click on “Register” on this page.

Stephen Wise has been designing electronic musical instruments since 1975. Steve specializes in realistic recreations of traditional instruments. He became interested in the field after hearing Walter/Wendy Carlos’ “Switched On Bach” and upon being introduced to the Allen Digital Computer Organ, the world’s first all digital musical instrument. Steve is currently designing instruments for Schulmerich Bells, maker of handbells, electronic carillons, and the breakout MelodyWave® instrument.