Power Amplifiers in Bridge Mode
First, some background information.
The maximum positive- and negative-voltage capability of a power amp is limited by the power supply's limits, i.e., its voltage rails. The amplifier can produce so many volts positive and negative at its output. The audio signal essentially swings between the positive and negative supplies within the power amplifier. These voltage rails are also referred to as the DC (Direct Current) supply. The DC supply in a CS®400X is +/- 52 Vdc; in a CS-800X, +/- 74 Vdc; in a CS-1000X, +/- 82 Vdc; and in a CS-1200X, +/- 86 Vdc. Each of these CS-X amplifiers can produce a continuous sustained sinusoidal signal level that is somewhere in the range of 55-58% below the rated value of its DC voltage rails. Power (in watts) is equal to the voltage produced by the amplifier times itself and divided by the rated loudspeaker load.
W = V x V/R
Now that the current Peavey CS-X series can deliver full rated power into a two-ohm load in stereo, each of these models can produce even more power in bridge mode into a four-ohm load.
When operating the CS-X series in mono bridge mode, you can think of the A channel as being the amplifier's positive voltage rail and the B channel as the negative voltage rail. Although this is not actually the case, it's convenient to simplify it this way, since the positive speaker terminal is connected to the A channel's red binding post, and the negative speaker lead is connected to the B channel's red binding post. In bridge-mono mode, no connection is made to the black binding posts on channel A or channel B.
As an example, let's look at how we can improve a typical sound system's performance by employing power amplifiers in bridge mode. Suppose your sound system included two Peavey SPTM 2XT loudspeaker enclosures powered by a single CS-800X amplifier. Each of the SP-2XTs are eight-ohm enclosures, and they would each receive 260 watts of power. If you are doing sound in mono, you could put the CS-800X in bridge-mono mode to power both SP-2XTs, connected in parallel across the amplifier's two red binding-post terminals. The CS-800X would now produce 1200 watts of total power or 600 watts to each of the SP-2XTs. You will now have an increase in SPL (Sound Pressure Level) in the order of +3.6 dB. Typically, you would have to double the number of enclosures to obtain a +3 dB increase in SPL. I just showed you how to obtain more than a +3 dB increase in performance for absolutely no increase in cost.
As another example, let's say you have either of these two systems: (a) two SP-4XTs and one CS-800X, or (b) two DTHTM 4s and one CS-1000X. If you added one more power amplifier to either of these two systems and operated both amps in bridge mode, you would basically have increased your financial investment by 33%, but your total available system power would increase by 285%, and your maximum SPL increase would be better than +4.5 dB.
If this interests you so far, let me explain further how bridge mode works so that you may understand an application, involving a three-loudspeaker array powered by a single CS-X. I stated above that you could think of the A channel as the positive voltage rail and the B channel as the negative voltage rail (Figure 1). Well, it's a little more complicated than that. To illustrate exactly what's going on, I am going to break each channel of a typical stereo amplifier down into two basic component stages. The first stage we will call the very first stage of amplification, and the second stage will be all the rest of the channels' components.
When the very first input stage amplifies the input signal, you have a certain amount of signal gain, as well as a 180-degree reversal in phase (polarity). When the amplifier is switched into bridge-mode operation, the signal at the output of the first stage of amplification of channel A is attenuated, (reduced in gain) so that it is at the exact level that was first inputted into the A channel, and it then becomes the B channel's input. If the first stage of channel A has a gain factor of 10, then the out-of-phase output of this first stage is attenuated to 1/10 its value. The bridge-mode switch then sends this out-of-phase signal into the input of the B channel. Now each channel is identical in level, yet opposite in polarity. If we hooked up a loudspeaker to channel A's output and another speaker to channel B's output, the two speakers would be moving in opposite directions, since the two channels are out of phase with each other.
In bridge-mode operation, however, we actually hook up the load between the two red terminals of both the A and B channels. While the A channel moves so many volts positive, the B channel moves the same number of volts negative. When the A channel swings negative, the B channel swings positive by the same amount of voltage. The loudspeaker doesn't know that it is hooked up between the two red binding posts, so it reacts to the difference in electrical potential (voltage) between its own red and black input terminals. When channel A moves +10 volts, channel B simultaneously moves -10 volts, and the loudspeaker sees a 20-volt difference in potential between its two terminals. In other words, the positive loudspeaker (red) terminal is 20 volts more positive than its negative (black) terminal.
Some people who don't understand bridge-mode operation put the amplifier in bridge mode and then hook up two loudspeakers, one on each channel. The two loudspeakers are now operating out of phase, and the bass response is considerably reduced, due to the two speakers opposing each other. Years ago, some power amplifiers were equipped with a switch that bridged (paralleled) the amplifiers' inputs. This allowed you to drive the two-channel amp monaurally with the same input signal, without patching the two channels' inputs together. However, this is not the case when operating a stereo amplifier in bridge-mono mode.
Now that you understand bridge-mode operation, I will discuss an application that employs a CS-X amplifier in bridge mode to drive three loudspeakers in a single array.
In rooms such as churches or auditoriums, where a single source of sound is desired, a common approach is to employ an array of loudspeakers flown directly over the front edge of the stage or sanctuary platform. For rooms that are twice as long as they are wide, a single loudspeaker system may not have adequate horizontal coverage to include all the seats to the near left and right in the pattern of the horn.
Modern loudspeaker systems employ constant-directivity high-frequency horns that exhibit uniform frequency response within their included angles of coverage. These constant-directivity (CD) horns are the best tools for the accurate reproduction of high-frequency information. However, these CD horns should not be placed side by side, where they can overlap in their coverage angles. If two of these CD horns are allowed to overlap a great deal, they will have diminished output directly on-axis between the two horns at some frequencies. Trapezoidal boxes are supposed to minimize the overlapping of these horn patterns. However, many of these boxes have trapezoidal angles that are smaller than the angles of coverage of their high-frequency constant-directivity horns. Therefore, if you pack the trapezoidal enclosures tightly, you still have severe overlapping of the high-frequency horns, resulting in diminished high-frequency output. Often the boxes must be splayed farther apart to minimize the overlap.
In auditorium applications where you need coverage for both the near and far seats, you can address the room with two sets of loudspeakers. The near seats are in the near field, and the far seats of course are in the far field. You can solve the problem of excess overlapping and still address the near- and far-field requirements by employing an array of three loudspeakers. In my example I am going to use three HDHTM 244T enclosures. The three speakers are flown in the center, right above the front edge of the stage or sanctuary platform. The center HDH-244T is flown right side up, with the horn on the top, and has some downward angle to it. The two outside HDH-244T loudspeakers are then flown upside down, with their horns on the bottom, and have more downward angle. With most Peavey enclosures, this downward angle is twenty to twenty-five degrees greater than the center loudspeaker.
In many rooms such as churches and auditoriums, where a single source of sound is desired, a common approach is to employ an array of loudspeakers flown directly over the front edge of the stage or sanctuary platform. Some rooms have an aspect ratio where they are twice as long as they are wide, and perhaps a single loudspeaker system will not have adequate horizontal coverage to include all the seats to the near left and right in the pattern of the horn.
Modern loudspeaker systems employ constant directivity high frequency horns that exhibit uniform frequency response within their included angles of coverage. The constant directivity, or CD horns, is the best tool for the accurate reproduction of high frequency information. However, these CD horns should not be placed side by side, where they can overlap in their coverage angles. If two of these CD horns are allowed to overlap a great deal, they will have diminished output directly on axis between the two horns at some frequencies. Part of the rationale for trapezoid boxes, is that they are supposed to minimize the overlapping of these horn patterns. However, it's a fact that many trapezoid boxes have trapezoidal angles that are smaller that the angles of coverage of their high frequency constant directivity horns.
Therefore, with many boxes, if you pack the trapezoidal enclosures tightly, you still have severe overlapping of the high frequency horns, resulting in diminished high frequency output directly on-axis. Often the boxes must be splayed farther apart to minimize the overlap of the horns.
In many auditorium applications where you need coverage for both the near and far field seats, you can address the room with two sets of loudspeakers. The near seats are in the near field and the seats of course are in the far field. You can solve the problem of excessive overlapping and still address the near and far field requirements by employing an array of three loudspeakers. In my example I am going to use three HDH-244T enclosures. The three speakers would be flown in the center, directly above the front edge of the stage or sanctuary platform. The center HDH-244T would be flown right side up, with the horn on the top, and it would have some downward angle to it. The two outside HDH-244T loudspeakers would then be flown upside down, with their horns on the bottom, and they would have more downward angle. With most of our Peavey enclosures, this downward angle would be twenty to twenty-five degrees greater than that the center loudspeaker.
Now the loudspeakers would be "Arrayed" properly, but we must do something to allow the center speaker to truly take care of the Far Field seats. The inverse square law says that the sound coming from a loudspeaker is reduced in level directly proportional to the inverse of the square of the distance away from the source. That sound coming directly from the loudspeaker is called the direct field. The inverse square law can be simplified if we relate it to the decibel scale. The direct field emanating from a loudspeaker drops in level or diminishes, at a rate of -6 dB, every time you double the distance away from the loudspeaker. Therefore in our application, in order for the Far Field loudspeaker to provide adequate coverage, the Near Field loudspeaker system must be turned down in level by -6 dB, so that the Far Field speakers can have a chance to provide the proper level to the farthest seats.
Also if the Near Field speakers are not reduced in level, the gain of the entire system, before feedback occurs, will be limited by the Near Field speakers themselves, because the Near Field speakers will react with the open microphones causing the system to feedback before the Far Field speakers can reach the level necessary to provide the desired SPL to the farthest seats.
In the past, if you wanted to use one power amplifier for this application, the two Near Field loudspeakers would be put on one channel of a power amplifier, and the gain or level control of that amp would be turned down -6 dB. The single Far Field speaker would be on its own channel and that channels level control would be set wide open. We would now have a situation where our gain structures would be set properly for the Near and Far Field components of the array. Suppose we were to use a CS-800X in this situation. When the Far Field loudspeaker received 200 Watts from the power amp, each of the Near Field speakers would received 50 Watts (-6 dB equals 1/4 Power). I am basing this exercise on the condition that each of the individual loudspeakers has an impedance of eight ohms. Two hundred Watts may or may not be adequate power depending on the room and the type of music to be reproduced. Also I would worry that some individual, who did not understand the need for limiting the level of the Near Field coverage loudspeakers by -6 dB, would come along and turn the power amps' Near Field channel up all the way. This would completely change the performance capability of the system as a whole.
I can show you how to get +6 dB more performance out of this system with no more financial investment, while having the system configured to prevent anyone from changing the calibration of the Near and Far Field gain structure. The secret to this application is to operate the CS-800X in bridge mode. Connect the Far Field HDH-244T across the two red binding posts of each channel, i.e., in normal bridge mode operation. Then you would connect one of the Near Field HDH-244T's, across the A channels' red and black binding posts. Next, the second Near Field HDH-244T would be connected across the B channels' red and black binding post; however, (note) the second Near Field HDH-244T, that is connected to the B channel output, would have the loudspeaker leads reversed. This is very important, since the B channel is out of phase with the A channel, you don't want the second Near Field speaker to be out of phase, so reversing the loudspeaker leads will maintain the proper polarity. In this application I would recommend reversing the loudspeaker leads at the speaker itself.
This would prevent someone from noticing the different color wires on the A & B channels' red and black banana posts, thinking they are "correcting" someone else's error by changing them around. In this application with a CS-800X, the Far Field HDH-244T would automatically have +6 dB more gain than the Near Field HDH-244T loudspeakers. When the Far Field loudspeaker receives 800 Watts from the power amplifier, each of the Near Field speakers would receive 200 Watts. I have shown you a way in this application to obtain +6 dB more performance in both the Near and Far Field, with no more investment in dollars. Typically, you would have had to quadruple the number of loudspeakers to obtain +6 dB more system performance.
The CS-800X can operate bridge mode into a four-ohm load. It can operate in stereo down to two ohms on each channel. In the above application we are connecting a single eight ohm loudspeaker across the two red binding posts of channels A & B, and then an additional eight-ohm speaker is connected to channel A, and another to channel B. As far as the amplifier is concerned, it's as if we had a four-ohm load hooked up in bridge mode.
Don't try to make this application anymore complicated than it already is. Many times when I have explained this approach to someone, they immediately try to figure out how to utilize even more loudspeakers in the same application. In some rooms, such as a slice of pie shaped auditorium or church, where the farthest distance can be to the left and right rear corners, the same three loudspeakers can be employed in a different configuration. That is, the center loudspeaker is flown upside down with more of a downward angle, while the two outside speakers are flown right side up with less of a downward angle. The center speaker is now covering the Near Field, so it requires the -6 dB of attenuation, and the two outside loudspeakers are now playing to the Far Field, so they do not need the -6 dB reduction in gain. With this type of array, there is no way to employ the above outlined application.
Here is one more application that can work, however, with the CS-X amplifiers. Perhaps we are in a room that is long and narrow, and yet we still need both Near and Far Field coverage without increasing the horizontal coverage angle. We can fly the two loudspeakers one above the other and configure the top (Far Field) speaker in the above bridge mode application, and then the bottom (Near Field) speaker can be connected across the A channels' red and black banana posts. This will work the same way as the first three-speaker array, i.e., the Near Field speaker will have the proper gain reduction (-6 dB).
I believe we covered a lot of ground in this article. If you thoroughly understand the operation of a power amplifier in bridge mode, you may want to undertake some of the outlined applications. If you are still a little cloudy on how bridge mode works, I would advise you not to try the above applications.
After I thought up this approach several months ago, I explained my proposed application to Jack Sondermeyer, Peavey Electronics' director of analog engineering. Since Jack is the engineer that actually designed the Peavey CS-X series of amplifiers, I knew that he would be able to spot any flaws in my thinking, regarding this application. After I explained this method to Jack he laughed a little, and I said, "On no, don't tell me there is a fly in the ointment"? Jack laughed once again, and said that he was laughing because it was a great idea, he couldn't believe that he hadn't thought of it.