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The Benefits and Advantages of DSP-based Audio
  by Bob Cunnings and Karl Winkler, Lectrosonics, Inc.

As DSP (Digital Signal Processing) has evolved in the past 10 years, remarkable new applications are now practical due to the immense power and speed of these devices. What was formerly only possible with large, complicated and expensive analog electronics can now be done with relatively inexpensive and compact DSP-based processors.

Economy, density and features

An important benefit of DSP technology is its ability to process many more channels or audio streams for a given cost than analog systems. DSP can make large numbers of analog components unnecessary, resulting in a simpler, more compact and less costly design.

For a small system with perhaps three audio channels or less, analog solutions can still be cost effective. Once there are a greater number of channels, however, or a desire to have the best performance from a system, the economies of scale can lean heavily towards DSP-based systems.

Another real benefit of DSP-based designs is the number of channels for a given rack size can be far greater than with an analog solution. Not only that, but the variety and number of processing steps can be greatly increased without increasing noise and distortion. Equalization, compression, limiting, delay and other types of processing can be applied to each channel at will, giving the user a great amount of freedom when setting up a DSP-based system.

For example, suppose you’re building a conference room that requires three channels of audio: a podium mic or perhaps a wireless lavaliere, plus stereo from a VCR or DVD player. You could set this up with an analog system, and it would require two mixers, two equalizers and a feedback eliminator to increase gain before feedback. But for about the same money, even a modest DSP system would give you these same functions and many more.

Once you have established your electrical design and applied a certain amount of DSP horsepower, you now how have a “blank canvas” to draw upon, in terms of what can be done to the signals. Analog systems rely on relatively fixed architecture, limiting the flexibility available to the user. In addition, the basic DSP engine can be part of a modular system, only requiring software to provide the specific functionality of each individual element.

Better with DSP

By handling the signal entirely in the digital domain, certain things can be accomplished that would be less precise, less repeatable or less reliable with an analog system. For instance, an almost infinite variety of filter types can be on tap, and new types could be added via software updates. You could use these filters for increasing gain before feedback, compensating for acoustical anomalies in the room, or just to shape the sound to taste. In addition, a greater level of precision can be attained by eliminating the variables that typically plague analog systems: thermal drift (i.e. that equipment parameters can vary due to temperature), component tolerances (precision analog components are available but expensive), and non-repeatability. Indeed, arriving at exactly the same performance from an analog system from day to day is not easy. With DSP, you can consistently call up the same settings as many times as needed, over whatever time frame necessary, and produce exactly the same results as before.

Not only this, but the results can have a higher level of quality with types of EQ curves or compression that would be very difficult to achieve in the analog domain. With today’s 32-bit floating-point architecture and accurate A/D and D/A converters, the performance of DSP-based systems is higher than that of the previous generations of equipment.

Only possible with DSP

While nearly impossible to implement precise delays with analog systems, it is relatively easy to do with DSP. This is a critical factor for any complex sound systems for acoustic wave alignment, or to re-align signals that have diverged, such as through video equipment. For instance, your audio may arrive early or late when compared to video from a hard drive file, and it will need to be re-aligned to avoid distracting your audience. In addition, certain types of filtering, such as phase-accurate EQ, are only possible in the digital domain. Beyond that, the range of filter types is nearly unlimited.

Time-based or dynamic processing such as compression and limiting can be handled in ways impossible in the analog domain. For example, you may have some presenters who wander on and off mic, where others speak quite consistently. The parameters of the compressor and limiter can be adjusted “on the fly” to compensate automatically for these differences.

Certain types of automation, such as auto microphone mixing, where microphone levels are automatically adjusted as people begin speaking, use sophisticated algorithms and can be done only with DSP. Digital processing has allowed for much more transparent mic mixing in conference rooms, boardrooms and in teleconferencing where smooth transitions to different input priorities is critical.

Scalability of DSP-based systems is nearly unlimited, where analog systems quickly begin to add noise and distortion to the signal as the length of the processing chain grows. Once the original signal has been digitized it can be manipulated, reproduced and routed without loss or noise buildup. Thus large-scale systems are entirely feasible with DSP, but very difficult with analog designs.

Because DSP-based systems run on software, or firmware, they can be updated simply by changing the software itself. Such updates allow for a longer operational life of the equipment before obsolescence, and they make upgrading far easier than in the past, when hardware components would have needed replacement. Maintenance, too, is much easier, since users are not tempted to try to “improve” the sound by tweaking settings. And since software can easily be transmitted via email or downloaded from web sites, access is easy and updates can be undertaken in the field.
 


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