When you talk over a conventional land-line phone (and many cellphones), the sound that you hear in your earpiece contains the caller’s voice,  plus a portion of your own voice mixed in (at “telephone quality”). It’s this “telephone quality” version of your voice that is the main focus of attention in phone hybrid equipment.

If there were no hybrid, one channel of your mixer would contain your voice at full studio quality. Meanwhile, another channel would contain the phone caller at phone quality, plus your voice at phone quality.

When the two versions of your voice (studio quality and phone quality) are mixed together, a distorted sound will result, as well as odd peaks and nulls at various frequencies.

Not only that… the low-quality version of your voice (the part we don’t want) is much LOUDER than the caller (that we do want). This makes it extremely difficult to mix the voices appropriately, since your voice is on both channels, and the caller is way too quiet.

The objective is to eliminate the poor-quality version of the talent voice. We want only the caller’s voice, with no trace of the talent voice. This is what hybrids attempt to do. As you might guess, low-cost hybrids don’t achieve this as well as the pricier ones. For under $200, analog hybrids like AUTOHYBRID and TT1 do a fair job. There is still a certain amount of low-quality talent voice remaining, still noticeable under the full-quality voice.

Digital hybrids like HOST, DH20, and HX-1 can almost completely eliminate the talent’s voice from the caller audio. They start around $500, and go up from there. Digital hybrids are the standard for broadcast-quality telephone calls.

Here is a link to our selection of telephone hybrids. As usual, each model will have its own unique assortment of features.

Analog hybrids will work fine for situations where audio is simply being sent down the phone line, or being received from the phone, with no 2-way conversation. However, if your need is for simultaneous 2-way conversation (like a phone interview), you should be looking at digital hybrids.

Hello? Hello?

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Not so fast there, bucko. For audio quality, digital is still a few (tiny) steps behind analog. Some very $$$ digital units are getting pretty close, but for the kinds of things that we can afford, analog audio quality is still better.

The area that digital has the advantage is transportability, and control.

A digital audio file can be copied a dozen times, and the 12th copy will sound exactly like the first. A 12th generation analog copy will have increased noise and poor signal-to-noise.

In the realm of control, digital can easily remember all of your settings. This is convenient for audio processors that need to behave differently for a sports program than for a classical music recital. One click, and everything changes instantly.

Digital mixers can remember complete “scenes”. Press one button, and all the faders, equalization, bus assignments, and effects are exactly where they need to be.

A digital audio item may be able to be controlled remotely over an IP connection, complete with visual display of the metering and other settings.

Is digital better? In some ways… yes. A large facility will realize the benefits of remote control, preset recall, and perfect audio copies. While a small, one-room studio may not need those features.

For practical purposes, digital audio and analog audio are both pro-quality. No significant difference. Feel confident in selecting the equipment based on the features it offers you, without fear that the choice of analog or digital audio will be audible.

Digital technology offers a lot of convenience. It’s not “better”… just different.

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But we sure do a lot of interview recording on portable machines. How do you choose which portable recorder is best suited for your use? Let me help…

First, let’s talk about the microphones. Some recorders come equipped with mics mounted to the outside of the case. This type of setup is best at collecting sound that originates across a wide area (environmental sounds, or a musical performance). For broadcast interviews, they mostly get in the way. Many models are available with mics mounted flush inside the case.

Even with flush mics, the recorder is still configured for a wide, stereo recording. The built-in mics will do in a pinch, but the preferred method is to use an external, handheld microphone for the interview. This gives you lower handling noise, more tightly focused pickup of your subject, and a place to hang a mic flag (promoting the station is always a good thing).

Pro handheld mics have XLR male connectors. Some of the recorders will accept XLR, while some use a stereo minijack. Fortunately, we stock the appropriate cables to connect to either style.

Next, let’s think about some of the special features that differentiate the recorders. We never record using “automatic” record level (you knew that, right…?) The record level will be constantly “searching”, and will distract from what your interview subject is trying to say. Instead, you should set the record level at the beginning, engage the limiter to protect against unusually loud blasts, and let it go.

Some recorders will monitor the sound level before you hit “record”, and determine the best level to use. This setting will be locked in when you begin the recording. Pretty slick.

Other units will take advantage of their stereo recording ability, and make a “backup” recording on track 2 at a slightly lower level (in case the primary recording on track 1 gets out of hand). Also pretty slick.

Of course, you will make your recording in a non-compressed PCM or WAV format. Only use MP2 or MP3 as a last resort, if you simply don’t have the available memory space to capture the event in PCM. This one step will make a significant difference to the quality of your piece, especially if it will be edited heavily before it is played on the air.

The recording is stored on a Compact Flash or SD memory card. Which style of card isn’t usually significant, as the card will probably never be removed from the recorder. The audio files are commonly transferred to the workstation over a USB connection.

There are digital recorders built into microphones, and adapters that will let you use pro mics with an i-phone or i-pad.

Think about the job you need to do, and the format that works best for you. Then have fun browsing the choices.

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See these two microphones? Other than the color, they look very much alike. One is priced at $99, the other at $325. It would seem that the $99 mic is just a cheaper copy of the $325 one, right?

Wrong.

These two mics couldn’t be more different:
One is dynamic, the other is condenser.
One is addressed from the side, the other from the end.

Which one is better? (here it comes…) It all depends on what you are planning to do with it. A dynamic mic is better suited for close-talking in a small room with less than ideal acoustics. A condenser is better suited for a singer or an acoustic instrument in a well-treated room.

Yes, you can use a condenser for speech, or a dynamic for music. The point is that you shouldn’t select a mic just because it looks nice. A pretty mic that doesn’t suit your needs is just a waste of time.

When considering a new microphone, think first about the job you want it to do. Speech, singing, instrument, quiet, loud? What about the pickup pattern? Does it need to be highly directional, a broad cardioid, figure-8, omni, or switchable? Then consider the acoustical environment it will be in.

Once you have narrowed the search to those mics that properly meet your needs, then you can start comparing them by looks and price. Too many mics are bought simply based on their appearance.

Take the time to determine the right type of mic, then have fun browsing.

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First, determine if the hum is being “heard” by a mic, and is simply being passed through the system as part of a valid audio signal. If it’s not part of the audio signal, hum is often caused by differences in the ground circuit on the various AC outlets you may have in use. Or perhaps that un-grounded power supply that runs your laptop computer. What’s the solution?

ART, Rolls, and Ebtech offer products (cleverly called “hum eliminators”) that will stop the offending hum without having any negative effect on audio quality. Once you have identified which device is creating the noise, simply connect its line level audio output cables to the hum eliminator, and from the other end of the hum eliminator to the original destination. The hum or buzz is now gone, because the internal transformers in the hum eliminator have isolated the electrical grounding discrepancy.

Hum eliminators won’t get rid of a hum that is actually part of the audio signal. They will only work on hum that is a result of an external problem, like the grounding issue mentioned above.

Hum eliminators are passive devices… no power is necessary. Most use 1/4″ TRS jacks for input and output. Some have a combination of RCA, XLR, and TRS jacks available. They are 2-channel units, so you can isolate a stereo device, or a couple of mono devices.

For larger setups, Ebtech makes a couple of rack-mounted models that give 8 channels of hum isolation: HE8 (1/4″ TRS), and HE8XLR (XLR). ART’s T8 has XLR, TRS, and RCA.

Stop that noise! I mean it!

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Most of our sales staff have been on the air (at the mic or at the controls) for decades.

Don’t get me wrong… our folks perform, mix, and record music too. But we know how to handle the spoken voice better than anyone.

We know why a $200 compressor may perform better for speech than a $2500 one.

We understand the difference between a phone patch and a talk-show phone system, and what it takes to make each one happen.

And microphones… we know that a mic that sounds terrific on a singer in a large recording studio may be completely wrong for speech in a small home studio.

Broadcast, voice-over and internet radio have very different needs. We can guide you to the best combination of equipment to get the job done with excellent audio quality, within your budget.

BSW is not a music store. We don’t sell instruments. We’re all about capturing the performance and delivering it to the listener, whether it’s a live venue or a radio broadcast.

We get it. We’ve got it. Call us. We’ll talk about talking.

800-426-8434
www.bswusa.com

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CAP-compliant gear must be in place by September 30, 2011. (the date has been extended to June 30, 2012.)

What’s this all about?

Under the existing EAS system, participants are assigned to monitor one or more local sources of EAS alert messages. This will be some combination of radio, TV, weather or public service broadcasts. When an alert is received, your EAS decoder determines if the message is intended for your coverage area, then switches the message to your transmitter.

The new CAP 1.2 requirement adds another monitoring source. This one is not over-the-air broadcast audio. It’s an internet connection. The CAP circuitry is built into current EAS encoder/decoder units from Sage and Digital Alert Systems. There are also add-on boxes that receive only the CAP information, and pass it on to your existing EAS gear. We carry CAP converters by TFT and Gorman-Redlich.

Your equipment will check the local CAP server frequently (every 30 seconds or so) to see if any alerts are available. When it finds one, it will be downloaded, and examined to determine if it applies to your coverage area. The CAP message will always contain text, and may additionally include pictures or mp3 audio files. If an mp3 audio file is present, it will be sent through your EAS gear, and out to the transmitter. If there is no mp3 audio file, the text will be converted to speech by your CAP unit, and the resulting audio will then be sent out.

CAP is not optional… if you are currently required to carry EAS alerts, you are required to be CAP-compliant by September 30, 2011 June 30, 2012.

Now you have a better understanding of what’s going on. Decide if you will add-on to your existing EAS gear, or replace it with a new all-in-one unit. Then give us a call. Don’t wait until the last minute… get your order in now. The clock is ticking.

Here are some links:
http://www.bswusa.com/eas.asp
http://www.bswusa.com/browse_subcat.asp?category=eassystems
http://www.bswusa.com/searchresult.asp?searchType=category&subcat=eas-ed

In live radio or internet streaming, there is no such thing as post-production. Everything happens in real time. Any manipulation of the sound – noise gating, equalization, compression, level balancing – is done by specialized pieces of hardware. Once the words are spoken, they are gone into the ether (or the “inter” I guess).  We’ve been doing it this way for quite a long time, and the equipment is pretty well established.

Is the process any different when the material isn’t “live”, but pre-recorded for later use?

Today, most audio editing programs have a lot of these functions available to apply to the recorded file. And, with a bit of skill and patience, you can correct many of the flaws of the original audio entirely in the software.

You will of course use software to cut out long pauses, breaths, coughs, etc. When that’s finished, you might run the audio file through several passes to normalize, equalize, compress, and limit the audio to make it “broadcast-ready”.

If you were using dedicated hardware for the EQ, compression, and limiting, those functions would not need to be duplicated in software, saving several minutes per file. Your editing session would simply be the cut-and-paste functions.

Even if you choose to do the final equalization, compression and limiting in software, it still makes sense to use a dedicated piece of hardware to process the microphone during the recording.

A mic processor’s downward expander function will do a more effective job of reducing pickup of the sound of your imperfect room. This task can be quite tedious when done in software (and nearly impossible once the tracks have been mixed together).

The mic processor’s compressor and de-esser will prevent clipping distortion on the recorded file. Once a file is clipped, it’s extremely difficult to undo the damage. Better to prevent it in the first place.

So, a hardware mic processor and hardware final compressor/limiter is the best combination for live programs. And pre-recorded programs should use a hardware mic processor, even if you elect to do the rest of the processing in software.

It’s about time. And time is money.

Here are links to some popular mic processors, program processors, and editing software:
Mic: 286S, 528E, CHANNEL
Program: 2M, OMNNIA1MULTICAST, VP-8PLUS
Software: SOUNDFORGE10, PROTOOLS, STUDIO-ONE-PRO

Out in the “regular” world, people wear headphones for all sorts of reasons…
- Don’t want to hear the other people around them.
- Want to listen loud, but don’t want to disturb the neighbors.
- Tunes while exercising.
- Want to be seen wearing the popular brand.

What seems most important is that they make a sound, not necessarily that they make great sound.

[tangent]
I’ve been watching reviews of consumer headphones. One that caught my attention recently talked about how the headphone in question had weak bass, inaccurate highs, distorted easily when turned up, was cheaply made and cost too much. But the reviewer still recommended buying them because they look so good.

I thought the whole idea was to get headphones that sound good? Maybe I’m just too old to understand anymore … I’ll just sit over here in the corner making bubbly noises in my glass of milk.
[/tangent]

In the pro audio world, headphones have specific jobs.
- In the on-air studio, headphones allow you to hear the program material while the microphone is on. A live speaker would create feedback. Any type that doesn’t “leak” sound into the microphone is OK.
- In the production room, a headphone can let you more clearly hear the fine details of your recording. Makes editing easier.
- A sportscaster uses headphones for noise isolation (so he can hear himself over the crowd noise), and to receive cues from his producer. Full-coverage is needed here.

Music mixing is one task that has traditionally been handled by the studio monitor speakers instead of headphones. Reference monitor speakers are carefully made to be as accurate as possible, so you can be sure that what you hear faithfully represents what’s actually on the recording.

Few headphones have the same sonic “signature” as a good pair of reference monitor speakers. Try it yourself. Select a very clean piece of music, set the speakers to a comfortable listening level, then match that level for the headphones. As you put on the headphones, the sound should ideally be nearly identical to the sound you heard from the speakers. You may be surprised by the difference.

Finding headphones with a similar character as your speakers has been a hit-or-miss adventure. The headphones that best represent the sound of the monitors are not always the most popular models. I have had good results with Audio-Technica ATHM40FS, Sennheiser HD280, and a few others. The new KRK KNS-6400 and KNS-8400 are specifically made to match the sound of KRK monitors.

When the headphones and speakers match, you can have confidence that your mix will be good whether you are wearing cans or not.

Now if you’ll excuse me, I need to go over here and make more bubbly noises in my milk. Rock on.

blup       blup        bbbbbbbbbb       blpp

First, let’s talk about the difference between analog and digital processors. In professional equipment, the sound quality is for all practical purposes, the same. Really.

Digital’s advantage is the ability to remember parameter settings (as a list of presets) and to instantly recall those settings. If you have only one delivery style, and only one microphone, you may never need to change the settings. However, if you are reading children’s books one day, and doing Monster Truck promos the next, it’s handy to be able to select the right settings with a single click.

The mic preamplifiers are clean, quiet, and robust, with plenty of gain for most any condensor or dynamic mic. (If you’ve got a high-end preamp that you “just gotta use,” it can usually be connected to the processor with little difficulty. Call us.)

The downward expander section will gracefully reduce the volume between words, and when you aren’t speaking. This will minimize pickup of the background noise in your room. Acoustic conditions are especially challenging in smaller rooms. If you don’t have an isolation booth, the expander can help a lot.

Before I go much further, I want to say a bit about settings. Yes, a mic processor has a lot of controls. That’s because it does several different jobs. There are usually no more than 2 or 3 controls for each task. Once you understand what each function does, the setup becomes easy. (Some folks are so intimidated by the numbers of controls, they figure “if I just turn everything up a little, it will be OK.”) If you don’t understand what a particular section does, bypass it until you have more information. Randomly turning knobs won’t get you where you want to go.

Now back to the article, already in progress…

The compressor section will be used more heavily by radio folks, much less for other VO tasks. In radio, the compressor can give the “in-your-face” loudness often needed for commercial spots and imaging work. Non-radio projects may need only a little compression, if any at all. (If your voice track is sent to someone else for inclusion in the finished project, use minimal compression. The client’s engineer will add compression if needed.)

Before we talk about equalization, consider this: You chose the microphone you are using because you like the way it sounds with your voice, and your delivery style. You probably listened to several before deciding this one had the right sound. With that said, there should be little reason to do a large amount of EQ. Start by listening carefully with the EQ disabled. If there is something missing, or a disturbing peak, apply just enough EQ to fix the problem. But not so much that you alter the sound of your carefully-selected mic.

Some mic processors have multi-band parametric equalizer sections. These are very powerful tools. Their operation is beyond the scope of this article. I’ll cover them in depth later.

And finally the “de-esser.” The name is a bit misleading. We don’t really want to completely remove the “ess” sounds. We just want to keep them in line. Some people make “ess” sounds very much louder than the rest of their speech. A “de-esser” is designed to limit the “esses” so they don’t stand out above the rest of the sound. If your “esses” are not excessive, ignore this section.

Don’t think of a mic processor as an “effect”, but as a “preconditioner”. When the sound leaves the mic processor, it is clean, loud, and under control.

Here are links to several of our most popular mic processors:
286S
528E
CHANNEL
2X
VORSIS-M1