Advanced Instrumentation and Digital Multi-Processing: A learning experience
Dec 1, 1997 12:00 PM, John Murray
At the fall 1992 AES convention in San Francisco, TOA demonstrated the remote control of the first DSP-based real-time signal processor in our industry, the SAORI, via a fiber-optic link. Having presented a paper on driver alignment using the SAORI and Tecron TEF-20 at the '93 AES convention in NY (Murray, 1993), I had dreamed of controlling both signal processing and acoustical analysis simultaneously via one computer. However, the control software for both devices were DOS-based, and MS-DOS couldn't run two simultaneously active programs that required a COM port each. What I really needed for site tunings was a mobile laptop PC, and only desk-top PCs could accommodate more than one COM port.
In the fall of 1993, I received my first copy of the beta software from engineering in Japan for the then-titled Integrated DSP and Control System. It was still MS-DOS-based, but the signal processing control capabilities were impressive. Later that winter, before the hardware was finished, I received the first Windows-based beta software for the system. Using Windows 3.1, I could run the signal processing software and then start up the latest version of TEF-20 Sound Lab software in a DOS window. Using "alt-tab" to toggle back and forth between the applications, I salivated about the future where I could make a settings change, hear it in real-time and measure the change graphically to see the acoustic results as well.
Spring of '94 brought the NSCA in Las Vegas and the unveiling of TOA's DACsysII series, as it is called here in the 'States. That summer, I received my first working prototype of the DP-0204 2-in by 4-out DSP unit. I had also recently began using a new laptop PC that had one COM port and two PCMCIA slots. I needed just one more item to realize my dream, and it came in the form of a serial I/O adapter in a PCMCIA card. This adapter finally gave me the two serial COM ports I needed to have both the TEF's acoustical analysis and the DACsysII's signal processing working simultaneously on one laptop PC. The addition of an RS-232C-to-RS-485 converter provided a balanced control line between the PC and up to 30 DSP units. This was another great leap ahead, because now I could tune the system in real-time from the middle of the listening environment, not from in front of the signal processing racks. No more endless stair climbs to and from the equipment room!
I had been tuning sound systems using TEF analysis since 1981. The original process then required an analog FFT analyzer, a spectrum analyzer, a high-quality signal generator with frequency counter, a black box interface from Richard Heyser and a scientific calculator to determine proper adjustments of the $40,000 test gear.
During my time at Electro-Voice, I had also spent much time in the anechoic chamber with the engineers on various loudspeaker development projects. It had taken two weeks to develop a passive crossover network used in a special DeltaMax loudspeaker produced for Mark IV's involvement at Euro-Disney in 1991. What I had now in '94 with DSP-based signal processing and analysis was light-years ahead. I could create an active crossover that worked much better in just 20 minutes while in my living room, and I could listen to the changes in real-time as I made them.
The total cost for this test set-up and all the needed signal processing, including a Toshiba laptop PC, Tecron TEF-20, TOA DP-0204, B&K 4007 calibrated test mic and all required software and cabling was roughly $11,500. This total is less than 30% of what was required in 1981 for the cost of the test equipment alone, and it performed much better and faster and was far easier to use.
The first problem I encountered using this set-up was how to use all the signal processing functions and how to properly interpret the resulting acoustical measurements. To my knowledge, no earlier system of acoustical analysis or signal processing had such power or ease of use. I had more tools at my finger-tips than was ever available before. Now that I had all these tools, what was the proper way to implement them all?
Avoiding acoustical contamination When developing the methodology described in the AES paper mentioned earlier, I had begun to realize that we as professionals in the sound business have no standard method to equalize sound systems. Regenerative (feedback tuning) and tuning to a particular RTA curve are two of the more common methods. And no matter which method is employed, much trial and error, adjusting and listening is always needed. Furthermore, because we generally use RTAs to equalize, we must interpret badly contaminated measurements during the equalization process. There must be a better way!
When the real-time analyzer was the most advanced measuring tool we had, Dr. Boner's "house curve" was the rather broad brush that accounted for the device's inability to distinguish between the anechoic, or what I will call direct response of a loudspeaker system, and what is commonly called the room response. What the RTA measures is a mix of the direct and power response, as affected by the reflective and absorptive nature of the environment. If the measuring mic is beyond critical distance, where the reverberant level exceeds the direct level, the power response dominates. With equal level in the direct response, the low-frequency driver, with less directional control, will have a greater power response than that of a better-controlled, beaming high-frequency horn. This is why the house curve viewed on an RTA has more level in the lower frequencies.
Because the RTA sees all the reflected energy as well as direct energy, a flat direct response will approximate the house curve with the test mic in the reverberant field. Of course, with varying pattern-controlled devices, the house curve was not very exact. One had to listen and adjust to get the sound "right." With the advent of non-beaming, pattern-controlled horns, their power response was more akin to the low-frequency components, therefore the high-frequency roll-off had to be less severe than the original house curve in order to get the sound "right." This method has served well for many years, but it is not exact and is time consuming. And it is still very subjective.
The method I employ uses a combination of proper microphone placement and time windowing to totally isolate the direct sound from contamination by any reflections or multiple sources. Placing the loudspeaker system outside will work, as will putting the test mic on a very tall stand if the loudspeaker system is flown. In other cases, as described in the AES paper, a combination of near-field mid and high measurements well within each transducer's coverage pattern, with a ground plane measurement for the lows, will work as well. The basic idea is to reduce the level of any reflections at the mic position to the point that they will not affect the direct sound at all. One should only equalize one transducer per passband of the loudspeaker system. This is especially true for higher frequencies (shorter wavelengths).
This technique enables the system tuner to quickly equalize for a flat direct response that produces, as near as possible at the point of the microphone, the same sound quality that exits the system mixing console. Whether one is sending the mix to a recorder or a loudspeaker system, it should make no difference in the mix if the loudspeakers are properly tuned. In recording studios, this is exactly the case when the manufacturer goes to great lengths to get a flat direct response from the main monitors.
In a sound-reinforcement system, the same can be true. If you initially equalize for a flat direct response, you will have the system 90% tuned right away. This method is hardly new, and if that is all there was to a perfect system tuning, the industry would have standardized on it long ago. However, for a sound-reinforcement loudspeaker system, other issues need compensation beyond merely tuning for a flat direct response. These issues account for the last 10% of the tuning, and their omission is what I think has lead people to think that this method is flawed.
The issues are as follows: Equalization to attenuate mutual low-frequency driver coupling for arrayed systems (one loudspeaker sounds great, an array sounds "tubby").
Knowing that room modes render meaningless the use of a test mic in a given space for the response below a critical frequency (f[subscript]c), regardless of the measurement system. This problem is caused by the position-dependent and greatly fluctuating levels encountered in the modal range of frequencies for a particular room (f[subscript]c in Hz = [3 x 1,128 feet/s] -- room's smallest dimension in feet).
Accounting for loudspeaker-to-measurement mic distance and the amount of high-frequency attenuation caused by air absorption of the direct sound level. (This issue deals with the fact that the human ear-brain combination finds a flat direct response to very high frequencies may sound intimate or "in your face" but unnatural to some listeners if the source's distance would normally produce an acoustic character with less high-frequency content).
Equalization to reduce high-Q mic-and-loudspeaker-supported resonances that are often too narrow to observe on some analyzers (those little colorations that hang on in time when you bark "check-one-two" into a system vocal mic).
Equalization in high-level systems to attenuate frequency ranges where a transducer's distortion components occur most strongly (can be synonymous with the last point).
Equalization incorporating some loudness contouring or "artistic EQ" (e.g. boosted rock'n'roll low-frequency "haystack" or a boosted 10 kHz range for "airy" vocals).
Often I have seen people try to equalize a system using a test mic on a standard floor stand on a hard floor in the reverberant field while playing pink noise over stereo stacks of multiple loudspeaker systems. They equalize for a flat response on the RTA's display. They are disgusted with what they hear when music is played over the system and assume RTAs are useless for equalizing. Except for approximating Boner's house curve and adjusting it until it sounds right, one cannot equalize a sound system that way.
Others use averaging, either of multiple response curves or of multiplexed measurement microphones. If your microphone positions are at null points at some frequencies, and in front of a system array they usually are, then you will be averaging good response with bad. Suppose all the mics happen to be at a null point for one frequency? If you could boost an EQ filter to +/-[infinity], would you?
In my opinion, to equalize a sound system quickly and accurately, one must isolate the direct sound. This means turning off all but one driver per passband and positioning the mic so that strong reflections either aren't there for an RTA to see or can be windowed out by the measurement system. With anything else, you cannot tell whether you are tuning the response of the loudspeaker or the effect of a delayed interference.
There are those who are of the opinion that one can perform room equalization. Don Davis, the founder of Syn-Aud-Con, originally coined this term. He has said, "If there's anything I'd like to take back, it's the term room equalization, because you can't equalize a room."
Let's look at the effect of a room reflection on a loudspeaker's frequency response. Because it is a time-domain effect, it is non-minimum phase and linear in nature. (See Figures 1 and 2.) This is (sin x)/x interference notching, popularly called a comb filter (even though it is not a filter and has nothing to do with a comb). Each notch in the comb-filtering effect that a reflection causes can be extremely deep, approaching infinite attenuation at the center, and is non-symmetrical on a log-scale frequency response graph. (See Figure 3.)
Conversely, equalizer filters are minimum-phase and symmetrical on a log scale, and I've yet to encounter any that have infinite boost at the center frequency. (See Figure 4.) Clearly, an equalizer's filters are not designed to correct notches caused by delayed reflections, let alone the multitude that exist at any test mic position relative to each loudspeaker in a sound system. Equalizers can only correct minimum-phase problems in the direct sound of a loudspeaker. (See Figure 5.)
Acoustical problems must be corrected acoustically. If a reflection off a wall is a problem, either re-aim the loudspeaker or add absorption to the surface. If a null at a particular frequency exists between two loudspeakers, re-aim, add a loudspeaker, try frequency shading, or live with it. You cannot fix lobing by averaging between good on-axis response and the nulls between devices. Any problem that is time-oriented cannot be fixed with equalization except at one single position in the room. This list includes every reflection in a room and any lobing caused by multiple sources. Techniques have been developed to address this, but they are very much a compromise. One must have a thorough understanding of both the measurement system and the resulting compromise to attempt this type of tuning.
Alignment and crossover networks Years ago I encountered UREI 813 Time-Aligned recording studio monitors. They had an Eminence 15-inch (381 mm) square-magnet subwoofer in an enclosure with an Altec 604 Dupex mid-high assembly consisting of a 1-inch (25.4 mm) compression driver on a very small 60 degrees X 40 degrees horn coaxially mounted through a 15-inch (381 mm) woofer. The entire system was passively crossed over via Ed Long' s patented Time-Alignment technique. The system sounded good for those days (late '70s and early '80s) as long as your listening position was slightly off-axis vertically. On-axis the horn was a bit overpowering. I also remember Don and Carolyn Davis demonstrating "signal alignment" by sliding a horn/driver assembly back and forth on top of a low-frequency assembly while pulsing the system. Ever since then, driver alignment has been a buzz-word in the industry.
When I first began experimenting with the SAORI, this was my first chance to really dig into driver alignment. I began asking people in the industry just how one went about aligning drivers. Delaying one driver's signal so that it arrived simultaneously with the other in the crossover was the first answer. However, this voice-coil alignment did not account for the phase shift introduced by the combination of the crossover filter in series with the loudspeaker as an acoustical/mechanical filter. The combination, more often then not, produced a dip in response at the crossover frequency that might be audible. (See Figure 6.)
That method conflicted with the phase-alignment shown to me by Jim Long using the XEQ-3 during my time at Electro-Voice. This technique adjusted the phase relationship between drivers at the crossover frequency to avoid the aforementioned response dip but did not account for different time arrivals. It involved reversing polarity of one driver, tuning for the greatest null at the crossover frequency using the delay all-pass filter control. Once this 180 degrees point was found (indicated by the deepest notch at crossover), the polarity would be un-reversed to be in-phase and flat through the crossover region using a 24 dB/octave Linkwitz-Riley network. (See Figure 7.) The technique, employed in the '93 AES paper referenced earlier, used the SAORI's digital delay to accomplish the phase alignment described earlier.
On Dec. 3, 1994, I presented another paper to the AES (Murray, 1994). This presentation was a live demonstration showing the creation of a crossover network and equalization for a small loudspeaker system using the DACsysII/TEF/laptop PC combination. The entire presentation, including explanations, took only about 35 minutes. To my knowledge, this was the first time real-time, simultaneous remotely computer-controlled signal processing and acoustical analysis was demonstrated in public.
The DP-0204 has the digital delay capability of the SAORI plus all-pass filtering like that offered by the phase-alignment-capable analog crossovers. I used the digital delay feature to align the woofer to the horn-driver by synchronizing the front edge of their respective broadband, full-range, unfiltered energy time curves. (See Figure 8.) Because their responses are unfiltered, the short wavelengths/high frequencies arrive first and are essentially voice-coil locators. This front-edge alignment synchronizes the voice coils of the woofer and horn-driver so that an unfiltered high-frequency impulse from either transducer reaches the measurement mic simultaneously. (See Figure 9.) This assures, even with additional crossover and EQ filtering, that the drivers' acoustic origins will be within a wavelength of each other at the crossover frequency.
Experience tuning systems with Craig Janssen has taught me that the next step in the process should be to equalize each driver separately before applying the crossover filters. If possible, the drivers should be equalized flat as much as an octave past the crossover frequency. Doing so makes combining the drivers via the crossover network easier because they act much more like the line-level signal from which crossover filter topologies are modeled. For example, if the loudspeakers are flat through crossover, after applying 24 dB/octave Linkwitz-Riley filtering, they will also be 6 dB down, just as a line-level signal would be. (See Figure 10.)
To do this equalization process, filters after the crossover on each output leg of the signal chain are essential. Any filtering that will affect the amplitude or phase of the signal past the crossover frequency into another driver's passband must use the filters located after the crossover. Systems with all the filters pre-crossover are useless for this, and those with only one or two filters after the crossover do not provide enough capability to properly tune most drivers.
Today's DSP-based units can provide a virtually unlimited number of possible crossover combinations. For choosing crossover slopes, the flexibility that DSP provides allows tricks that were not possible when only relatively rigid, symmetrical, analog crossovers were available. One must guard against "rapture of the deep" in searching for the perfect combination. You can rapidly use up all the time that computer-controlled tuning is supposed to save you!
This article is not the proper venue for an in-depth discussion on crossover filtering, but I can offer the following food for thought. If the chosen crossover slopes provided only a 3 dB down-point at crossover, such as an 18 dB/octave Butterworth function, a 3 dB hump at crossover would occur if the drivers were in phase at that frequency before the crossover filtering was applied. You could choose to spread the frequency hinge-points by moving the low-frequency crossover frequency down and the high-frequency frequency up so that the hump disappears. Or these hinge-points could be chosen so that each driver's level is 6 dB down at the chosen crossover frequency. Then an APF or delay could be used to provide an in-phase summation and a flat response will result regardless of the filter type employed.
Although most people I've worked with choose 24 dB/octave Linkwitz-Riley slopes, other filter types also offer attributes. For example, Bessel filters exhibit minimal group delay compared to other more commonly used Butterworth and Linkwitz-Riley topologies.
One should never lose sight of the fact that the acoustical crossover slopes and frequencies measured are rarely those chosen electrically in the crossover network. The mechanical filters (loudspeakers) after the amplifiers change the signal characteristics. It is the acoustical result that is important and to which we listen.
Because of the mechanical filtering, the acoustical product usually does not closely mimic the amplitude and phase characteristics of a line-level crossover filter. As a result, proper summation won't exist at crossover acoustically. One can either apply an all-pass filter (APF) or readjust the digital delay to one of the drivers to phase-align them.
When using the APF, set it to the lowest possible Q so that its effect is smooth and gradual to avoid near-crossover phase cancellations with the other driver's idiosyncrasies. When using delay for this, be sure the summation point has the drivers within a wavelength at the crossover frequency. In-phase summation at crossover avoids a deep notch in the frequency response and provides a smooth phase-response transition through the crossover region. The most desirable result exhibits the least phase shift from the low frequencies to the high frequencies.
Keep in mind that alignment between drastically different wavelengths, such as 100 Hz at 11.3 feet (3.4 m) and 10 kHz at 0.113 feet (0.03 m), is fairly academic. If perfectly aligned, the 10 kHz wave will go through full cycle long before the 100 Hz wave even begins to rise in level.
Using this method of voice-coil/impulse alignment-equalization-crossover-phase alignment, I've easily tuned many combinations of drivers to within a +/- 2 dB window, with a smooth phase response, throughout the entire device's passband on axis. (See Figure 11.) Given reasonable choices, this is virtually regardless of the transducers employed.
For single-device applications, this on-axis frequency response curve may be averaged with off-axis curves within a device's coverage pattern. However, if drastic response-curve differences are encountered within the published coverage angles of the device, go with a flat on-axis direct response and change your product selection next time. In applications employing multiple transducers per passband, I would recommend tuning only the on-axis response. Any off-axis averaging would be compromised by other adjacent sources in the system.
The future First, one must keep in mind that the function all these DSP-powered parametric filters, high-pass and low-pass crossovers, digital delays, all-pass filters and compressor-limiters are solely for a loudspeaker system. If the DSP function occurs post-mixer, its purpose is to properly route to, protect, or correct for a loudspeaker. We must always think in terms of the effect on the loudspeakers when we employ all these tools.
When the DACsysII series was first introduced, the ability to mimic a 1/3-octave equalizer was a high priority. Most people at that time were using 1/3-octave EQs in most systems, and FFT-based measurement systems, such as Tecron's TEF or Meyer's SIM system, were not in the majority. RTAs were, by far, the industry's standard method of acoustical analysis. As a result, analog parametric equalizers were not widely used because of the difficulty in documenting their settings and the RTAs inability to resolve their adjustment parameters.
RTAs are still the most numerous, but Sam Berkow's economical SMAART system, offered by JBL, is quickly gaining popularity, and soon FFT measurement systems like this will constitute the industry's more cost-effective standard. Because of the ability of these types of measurement systems to resolve their adjustments, and because of the ability of the DSP-based filters to have their settings documented, the current trend is a much wider industry acceptance of parametric equalization.
I believe multiple parametric filter sets located after the crossover network on each output leg in the system signal chain will be the trend for future DSP products. Each of these filter sets will feed an individual driver type within a system. Once you have tuned a system using filters after the crossover rather than before it, you realize that after the crossover is the proper place for equalization. Historically, equalization has always been in front of the crossover simply because it was more economical to assemble all the filters in one box, and this dictated a position in front of the crossover. The flexibility of DSP has freed us of that convention, and the industry will be drawn in the opposite direction from now on.
Companies offering DSP products in our industry are coming to realize that they are software companies and that the control software for their products is the product from the purchaser's point of view. As a result, much better graphical user interfaces (GUIs) will be available for PC-controlled products in the near future. "Better" means designed from the user's rather than the programmer's point of view. Unfortunately, the fixed-installation industry is moving away from the touring industry's MAC use, but for PC users, things are only going to get better. Additionally, with the availability of spread-spectrum wireless RS-232C links, wireless control will soon be the norm. For sound system tuning, ease of product use will soon advance phenomenally.
Since the first time I experienced simultaneous signal analysis and processing, I have been dreaming of the day when acoustical analysis would be incorporated into signal processing hardware and control software. Today we have both analysis and processing software operating under Windows 3.1/3.11 and Windows 95. Perhaps by the time you read this, a combined product will already be on the market. Its time is here.
The crossover networks I've constructed above are, to a great extent, synonymous with their analog counterparts. Their propagation delay is relatively short because of the infinite impulse response (IIR)-based digital filtering. IIR and analog crossover filtering have a predictable phase shift from the low-frequency section to the high-frequency section. For example, a 24 dB/octave Linkwitz-Riley crossover with some equalization filtering will have more than 360 degrees of phase lag from the lows to the highs above the crossover frequency. Other filter functions can provide equally flat frequency response while having less phase shift. Bessel filters, for example, can provide a flat response with only 100 degrees of phase shift from the lows to the highs.
Finite impulse response (FIR)-based DSP or use of multiple APFs are capable of providing essentially a flat phase response from the first break-up mode of the high-frequency drivers in the 10 to 20 kHz range down to subwoofer frequencies. Although flat phase response does sound subtly better, the price to be paid is a relatively long propagation delay. Depending on how low in frequency the flat phase response extends, the delay from input time of the signal processor to output time can be 30 ms or more. Also, the present techniques to apply the tools for flat phase, FIR filtering and multiple APFs are sufficiently difficult and time-consuming to keep these tools from being an option in the field at this time.
Some products in the marketplace now attempt automatic equalization. Though these products are not nearly sophisticated enough to be of use professionally, we will soon see products that are. Once we see acoustic analysis modules incorporated into the more powerful DSP products we currently have, the market will be ready to take the next step, adaptive filtering. This is a more intelligent version of automatic equalization that will use phase-sensitive, FFT-based measurements to align and equalize sound systems, providing a preset compromise between flat-phase and maximum allowable propagation delay.
As outlined by Rob Reams, the creator of Audio Control's Iasys, these systems will look at parameters such as maximum level before non-linearity, ambient noise level and highest and lowest frequency of useful reproduction. These types of systems will then isolate areas of non-minimum phase response. Once these types of information are incorporated into the intelligent control software, the adaptive filtering can begin. This complex process involves a microprocessor that controls DSP to adjust parameters until a target response at a test mic is realized. We have all the technology to do this now, and some sophisticated adaptive filtering systems do exist, but they are cost-prohibitive and do not have a GUI suitable for the general market yet. This will soon change.
Imagine setting up a sound system, placing test mics for each particular portion of the system's response curve, speaking "equalize" into your laptop's built-in mic, and getting a cup of coffee. When you return, your system is perfectly tuned. Now imagine your customers doing this, too. Imagine what job opportunities you'll have once this happens. Hmmmm ... ever thought about a career in lighting?
Author's note: I would like to thank all those who have taught me everything I know about audio, including the individuals mentioned in the body of this text and those who were not. Most everything in this article has been taught by, or stolen from, others in this industry that I admire.
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