DSP Processing For Sound Systems

In Uncategorized by tfwm

History

The sound system for any house of worship is often the focal point of planning, discussion, revamping and re-engineering. After all, being able to communicate a powerful message to an ever-growing audience will always be the primary goal. To that end, the sound system has almost always required the specialties of acoustic and sound engineers and consultants to take the goals of the parish and convert them into an affordable, operable and high-performance designs.

In many parts of the world, but North America especially, worship services have pushed the “edge of the envelope” in respect to what is being asked of the sound system, to larger and larger venues and to an ever increasing sound pressure level. Today, it is not uncommon to outfit a 4,000-seat auditorium with a concert quality sound system. Not only must you amplify the pastor’s voice, now you must support the 12-piece gospel band, (with full monitors), amplify the choir and send processed signal to satellite rooms for the overflow! A far cry from the one microphone and choir in a 200-seat parish.

Along the way, sound system design has taken some radical leaps forward in technology. Where analog mixers and simple processing like EQ were done in “all-in-one” system solutions, or perhaps discrete components, today’s DSP (digital signal processing) options give the system owner/user greater options in terms of design, flexibility, processing power, performance and future expandability.

To many, the operation of a simple sound system might seem intimidating, so it is easy to see how a DSP sound system might overwhelm the average owner/operator. It is the intent of this three-part series to describe the benefits, explore the options and increase the comfort zone when it comes to specifying and enjoying your next worship sound system.

Before DSP

In the simplest sound system, one or more microphones are taken to a mixer and these signals are summed together and delivered to an amplifier/speaker combination. Although this seems simple enough, more often than not, the sound system is compromised in many ways. Some problems are not related to equipment, such as placement of the speakers relative to the microphones (your speakers are out in front, right? Think gain before feedback). Other factors include efficiency and size of the speakers (pattern, directionality, power handling, bi-amplified etc.). Upon examining the microphone, items like polar pattern and sensitivity, rejection of handling noise etc. come into play. And then there is the mixer….(illustration)

In our simple example, the mixer seems innocent enough but there are plenty of things to be improved upon. First, the microphone input must be of a high enough quality to amplify the microphone signal while imparting the minimum noise. Also, phantom power for those condenser microphones would be nice. In most cases, the EQ section of the smallest mixer is typically over used. Lastly, if there is any output EQ, it is probably broad in nature, suitable for general tonal adjustment and not designed for “surgical” precision like taking out feedback or fixing room anomalies.

Now, having looked at the simplest analog system, let’s take a look at how a DSP advance can help. In regards to the DSP mixer, some devices available today give you the same high quality mic pre-amplification that is available on larger, professional mixers, and as a bonus, are programmable and resettable; including the phantom power status! Mixing can be done manually, of course, but you can also preset the mixer for a variety of functions. Many automixing algorithms are available, each using a variation of functional principals from simple gating of channels to complex gain-sharing methods. The channel EQ on the old analog mixer is again replaced by DSP and can be preset for a variety of users. The output of the mixer can feed additional EQ, which addresses the room, feedback nodes and other anomalies. Also, digital crossovers can be used to improve the efficiency and assure that each speaker is receiving the right EQ at the proper level. There is even DSP available to compensate for increasing or decreasing noise in the room, which adjusts the overall level automatically. Some DSP based systems can even monitor the status of all the amplifiers. (illustration: previous page)

So, if you want a high-tech, re-settable, programmable, tamper-resistant sound system, DSP is your answer and you should read on!

What is DSP?

As mentioned earlier, DSP stands for Digital Signal Processing. But that means something even bigger. First, our audio signal comes out analog; either from a microphone or an instrument. So, it must be converted to digital in order for us to perform any digital processing to it. We are mostly familiar with the conversion due to other digital media like DAT or recordable CD. Once the signal has been converted, the “bit stream” can be manipulated to perform different tasks. A simple task, like changing the level, is accomplished by adjusting the “numbers” (the digital bitstream) by a certain ratio. The calculations are performed in the DSP chip. The task that the DSP performs is called an algorithm. Once the digital signal completes the instructions in the DSP, it goes on to the D/A (digital to analog) converter and is output to our amplifiers and speakers.

In today’s DSP audio products, it is possible to perform many operations or processes on a given signal. These chips are a programmed part of a DSP audio component. Some devices are fixed to perform a specific task, while others are programmable by the contractor/user to more appropriately adapt to the unique requirements of a given installation. More sophisticated changes in level, like compression, limiting, gating etc. are similar, each with specific detectors and gain shaping functions to fulfill the job.

The power of a DSP chip is measured in MIPS or “millions of instructions per second”. The bigger the number, the more processing the chip can perform. The fact that DSP is measured in terms of instructions to be performed in a given time brings up another interesting point – DSP takes time. This time is often referred to as latency. In fact, there are a few causes for small delays due to a DSP based sound system. First, the analog to digital to analog conversion (A/D/A) takes an amount of time (acquisition and reconstruction). This time is a fixed component of the converters and usually adds up to less than 3 milliseconds. The latency due to the DSP can vary depending on the number of processes being done and the number of “chips” being used. The range is usually within 2 to 10 milliseconds with the faster DSP chips doing more processing in the shortest amount of time. This might seem insignificant except that, in most performance systems, you need to listen to playback (monitor) from the mixed source. If this mixed source is adding a large amount of delay due to processing, the foldback will not come back to the performer in time. Generally speaking, the higher quality processors have worked out these kinks, but there are a few on the market that offer decreased usability when put to this test.

Why DSP anyway?

The trend towards DSP based audio systems is being driven by a number of factors. First, sound system design can be made system specific – each venue having a unique processing path that serves a specific purpose. In addition, a flexible DSP path can be changed and modified to experiment with new processing or to accommodate changes in the venue. Sonically, a sound system can benefit because the more integration of processing in the digital domain can yield a far better dynamic range than can be achieved with discrete analog components. Further, many wiring and interconnect errors or logistical failures are eliminated. Typically, rack space is reduced as much more processing can be done in a far smaller rack than the analog equivalent. And, although there is the possibility for a DSP audio system to be more complex, careful programming by the contractor can make it easier to operate for the user. Many settings can be “cloaked” or restricted from view, leaving only pertinent ones accessible. Lastly, new processing concepts can be added via software updates allowing the hardware to stay current with industry advancements.

How many times has your parish just wanted to return the sound system to exactly the way the sound technician left it? In analog, this is nearly impossible. But, since DSP sound systems are programmable, they are also, by nature, resettable. This is a major issue in many venues; especially where many different styles of services are being held over the day and throughout the week. These presets or “scenes” can be stored and recalled for a service later. Also, many companies have included multi-level security to help filter out unauthorized adjustment and, in some cases, render the system completely tamper proof.

Processing Types

Almost any processing type available in analog can be performed in DSP. In some cases, DSP is actually a better solution because it requires less support and interface hardware. From mixers to EQ, compressors and gates to ambient noise sensing and feedback reduction to crossovers, all major audio processing can be done in DSP.

In some cases, the choice for DSP is purely economical. For instance, the most complicated automatic mixer requires many additional components for detection, filtering and adjustment of the input levels. In DSP, all these components are modeled as part of the algorithm. Some optional processes can even be switched in and out to conserve the DSP.

Take the example of equalization. The 30 band graphic EQ is the “meat and potatoes” of nearly every analog sound system. Largely favored for its interface, graphic EQ provides the user with a “picture” of the curve it is processing. But, drawbacks in the analog version such as many moving parts, many amplifier stages, noise and distortion are among the major detractions. In DSP, providing the user with 30 filters is possible, but expensive due to each filter requiring a separate instruction. It is more ideal to use a parametric equalizer, which most engineers favor for sonic quality, in order to consume fewer filters and therefore less DSP processes. In some cases, this can yield an efficiency of up to 60% of a DSP graphic type. In addition, the interface (usually on a computer screen or LCD display) can display exactly what the EQ is doing; including the interaction between bands, rather than an approximation, which is what you get with analog.

In the next part of the series, we’ll look at the virtues of fixed and flexible DSP layouts, control of DSP audio systems and networking of digital audio signals.