by Rick Allen

Digital audio! It has opened a whole new world of possibilities in radio production: precision editing, the chance to undo any edits you didn't like, the courage and time to try some edits you would have never attempted before, no tape hiss or generation loss when making copies. The list goes on. But then we discovered that despite these advantages and the fact that newer digital audio systems have become pretty straightforward, we still hit a snag with some technical problem every once in a while, and it always seems to happen when the PD is screaming for that promo he wants to run before the next stop set. So here are some tips and concepts for general digital audio recording and a few more that apply to the lucky group of us who use it for our jobs in radio. For some people, I know this look at digital recording will be a basic recap of areas you're already very familiar with. For others, it will seem very close to a foreign language loaded with too many numbers. Hopefully though, if you're like me, every time you learn something new about this great technology, you find several more areas you want to explore.

On a very basic level, the same principles of digital audio apply to the recording, the playback and the storage of sounds. This includes DATs, MiniDiscs, CDs, samplers, hard disk recorders and digital multitrack tape machines. It all begins by converting the analog audio (waveform) into numbers that represent the sound in a way that a computer can use to capture and manipulate the audio. This is done by measuring the amplitude and frequencies of the incoming signal at a predetermined rate and converting them into a binary number.

Two factors of this representation can drastically affect the quality of digital sound. The first is the sample rate. Your CD player plays back at 44.1 kHz. Your DAT recorder might record at 44.1 or 48 kHz. This rate is the number of times each second that the signal is measured. The faster these measurements are taken, the more the numerical picture represents the original waveform, and the higher the frequency of the audio that can be captured and played back. However, you'll also be eating up more hard drive space with bigger chunks of numbers. The frequency that corresponds to half of your sampling rate is the maximum frequency that your system can reproduce. So, if you're sampling at 48 kHz, you can capture audio between zero and 24 kHz. Remember that FM broadcasting's bandwidth drops off sharply at 15 kHz. So if you want more recording time, you can sample at 32 kHz (giving you audio up to 16 kHz). To give you a reference, a 2-gig hard drive can record around 531 track minutes at a sampling rate of 32 kHz, 385 minutes at 44.1 kHz, or 354 minutes at 48 kHz. Choosing 32 over 48 gets you 146 extra minutes of recording time. On the other hand, remember that once you record at the lower sample rate, you can never recreate those higher frequencies. This came back to haunt me at HOT 97 once when I did a quick in-house edit of a song at 32 kHz. The record company liked it and asked my PD for a copy so they could release it on a CD. At 32K, the edit sounded just fine on the air, but everything above 16 kHz would be missing on a CD. I had to go back and recreate all my edits a second time after resampling the original song at 44.1 kHz. A lot of extra work just because I wanted to save some hard disk real estate.

The other factor is resolution. This is the number of bits used to represent each signal measurement. You've heard of 8-bit samplers or 16-bit CD players. There are many choices out there. What's the difference? Well, a string of 8 binary numbers (8-bit) can express 256 different signal levels when recording a snapshot of audio, while a 16 bit system can represent 65,536 different levels. The higher the resolution the closer the number represents the true level of the analog signal, but again, the more space you need to store it.

Another factor comes into play once you've recorded audio and are doing a digital to digital transfer. You have to pay attention to the clock source. The clock gives both digital systems the timing reference so all the information is placed at the same place on the copy as it was on the original. Some systems choose the clock source automatically. If you're working with one that doesn't, you must tell the destination system to listen to an external clock (the clock from the original source). It is possible to get clock feedback loops. If this happens, recheck and make sure the only digital connection plugged in is the source digital output into the destination digital input.

That's a very simple look at just a few areas of the digital domain. Don't let any of the technical jargon get you frustrated. Always remember, digital audio is not brain surgery. I can assure you it is just as complicated and risky, but it is always a lot more fun, and most of the time there's a lot less blood.

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