by Major Venudhar Singh, Research Associate (Sur.Aur.Saaz Team)

Audio Interconnections

Q1 - How are professional transmission lines and levels different from consumer lines and levels? What is -10 and +4? What's a balanced or differential line?

Professional transmission lines differ from consumer lines in two ways. First, consumer lines tend to run about 14 dB lower in level than pro lines. Second, professional lines run in differential, or balanced, configuration. In a single-ended line, the signal travels down one conductor and returns along a shield. This is the simplest form of audio transmission, since it is essentially the same AC circuit you learned about in high-school physics. The problem here is that any noise or interference that creeps into the line will simply get added to the signal and you'll be stuck with it. In a differential line, there are three conductors. A shield, a normal "hot" lead, and a third lead called the "cold" or "inverting" lead, which carries a 180-degree inverted copy of the hot lead. Any interference that creeps into the cable thus affects both the hot and cold leads equally. At the receiving end, the hot and cold leads are summed using a differential amplifier, and any interference that has entered the circuit (called "common-mode information" since it is common to both the hot and cold leads), gets canceled out. Differential lines are thus better suited for long runs, or for situations where noise or interference may be a factor.

Q2 - What is meant by "impedance matching"? How is it done? Why is it necessary?

We can talk about the characteristic impedance of an input, which is to say the ratio of voltage to current that it likes to see, or how much it loads down a source. (You can think of this as being an "AC resistance" and you would be mostly right, although it's actually the absolute magnitude of the vector drawn by the resistive and reactive load components. Dealing with line level signals, reactive components are going to be negligible, though).

In general, in this modern world, most equipment has a low impedance output, going into relatively high impedance input. This wastes some amount of power, but because electricity is cheap and it's possible to build low-Z outputs easily today, this is not a big deal.

With microphones, it _is_ a big deal, because the signal levels are very low, and the drive ability poor. As a result, we try and get the best efficiency possible from microphones to get the lowest noise floor. This is often done by using transformers to step up the voltage or step it down, to go into a higher or lower Z load. Transformers have some major disadvantages in that they can be significant sources of nonlinearity, but back in the days of tubes they were the only solution. Tubes have a very high-Z input, and building balanced inputs with tubes requires three devices instead of one. As a result, all mike preamps would have a 600 ohm balanced input, with a transformer, driving a preamp tube. Today, transistor circuits can be used for impedance matching, although they are often more costly and can be noisier in cases.

As a result of the expense, consumer equipment was built with high-Z microphone inputs, and high-Z microphones. This resulted in more noise pickup problems, but was cheaper to make. Unfortunately this still held on into the modern day of the transistor, and a lot of high-Z consumer gear exists. Guitar pickups are generally high-Z devices, and require a direct box to reduce the impedance so that they can go into a standard 600 ohm mike preamp directly.

Many years ago, the techniques that were used in audio came originally from telephone company practice. Phone systems operate with 150 or 600 ohm balanced lines, and adoption of this practice into the audio industry caused those standards to be used. In the modern age where lines are relatively short and transformers considered problematic, the tendency has been to have low-Z outputs for all line level devices, driving high-Z inputs. While this is not the most efficient system, it is relatively foolproof, and appears on most consumer equipment. A substantial amount of professional gear, however, still uses internal balancing transformers or resistor networks to match to a perfect 600 ohm impedance. Modern equipment works on principles of voltage transfer rather than power transfer. Thus a standard audio circuit today is essentially a glorified voltage divider. You have a very low output impedance and a very high input impedance such that the most voltage is dropped across the load. This is not an impedance-matched circuit in the classic sense of the word. Rather, it is a "bridged" or "constant voltage" impedance match, and is the paradigm on which nearly all audio circuits operate nowadays.

Q3 - What is the difference between dBv, dBu, dBV, dBm, dB SPL, and plain old dB? Why not just use regular voltage and power measurements?

Our ears respond logarithmically to increases in sound pressure level. In order to simplify the calculations of these levels, as well as the electrical equivalents of them in audio systems, the industry uses a logarithmic system to denote the values. Specifically, the decibel is used to denote logarithmic level above a given reference. For instance, when measuring sound pressure level, the basic reference against which we take measurements is the threshold of hearing for the average individual, 10^-12 W/m^2. The formula for dB SPL then becomes:

10 Log X / 10^-12 where X is the intensity in W/m^2

The first people who were concerned about transmitting audio over wires were, of course, the telephone company. Thanks to Ma Bell we have a bunch of other decibel measurements. We can use the decibel to measure electrical power as well. In this case, the formula is referenced to 1 milliwatt in the denominator, and the unit is dBm. 1 milliwatt was chosen as the canonical reference by Ma Bell. Since P=V^2 / R, we can also express not only power gain in dB but also voltage gain. In this case the equation changes a bit, since we have the ^2 exponent. When we take the logarithm, the exponent comes around into the coefficient, making our voltage formula 20 log. In the voltage scenario, the reference value becomes 0.775 V (the voltage drop across 600 ohms that results in 1 mW of power). The voltage measurement unit is dBv.

The Europeans, not having any need to abide by Ma Bell's choice for a canonical value, chose 1V as their reference, and this is reflected as dBV instead of dBv. To avoid confusion, the Europeans write the American dBv as dBu. Confused yet?

Q4 - Which is it for XLRs? Pin 2 hot? Or pin 3 hot?

Depends on whom you ask! Over the years, different manufacturers have adopted varying standards of pin 2 hot and pin 3 hot (and once in a while, pin *1* hot!). But nowadays most manufacturers have adopted pin 2 hot. Still, it is worth taking the extra minute or two to check the manual. The current AES standard is pin 2 hot.

Q5 - What is phantom power? What is T-power?

Condenser microphones have internal electronics that need power to operate. Early condenser microphones were powered by batteries, or separate power supplies using multi-conductor cables. In the late 1960's, German microphone manufacturers developed 2 methods of sending power on the same wires that carry the signal from the microphone.

The more common of these methods is called "phantom power" and is covered by DIN spec 45596. The positive terminal of a power supply is connected through resistors to both signal leads of a balanced microphone, and the negative terminal is connected to ground. 48 volts is the preferred value, with 6800 ohm resistors in each leg of the circuit, but lower voltages and lower resistor values are also used. The precise value of the resistors is not too critical, but the two resistors must be matched within 0.4%.

Phantom power has the advantage that a dynamic or ribbon mic may be plugged in to a phantom powered microphone input and operate without damage, and a phantom powered mic can be plugged in to the same input and receive power. The only hazard is that in case of a shorted microphone cable, or certain old microphones having a grounded center tap output, current can flow through the microphone, damaging it. It's a good idea anyway to check cables regularly to see that there are no shorts between any of the pins, and the few ribbon or dynamic microphones with any circuit connection to ground can be identified and not used with phantom power.

T-power (short for Tonaderspeisung, also called AB or parallel power, and covered by DIN spec 45595) was developed for portable applications, and is still common in film sound equipment. T-power is usually 12 volts, and the power is connected across the balanced pair through 180 ohm resistors. Only T-power mics may be connected to T-power inputs; dynamic or ribbon mics may be damaged and phantom powered mics will not operate properly.

Q6 - How do you interconnect balanced and unbalanced components?

First, let's define what the terms mean. The simplest audio circuit uses a single wire to carry the signal; the return path, which is needed for current to flow in the wire, is provided through a ground connection, usually through a shield around the wire. This system, called unbalanced transmission, is very susceptible to hum pickup and cannot be used for low level signals, like audio, for more than a few feet. Balanced transmission occurs when two separate and symmetrical wires are used to carry the signal. A balanced input is sensitive only to voltage that appears between the two input terminals; signals from one terminal to ground are canceled by the circuit.

The simplest way to connect between balanced and unbalanced equipment is to use a transformer. The signals are magnetically coupled through the core of the transformer and either side may be balanced or unbalanced. Good transformers are expensive, however, and there are cheaper methods that can be used in some instances.

An unbalanced output can be connected to a balanced input. For instance, from the unbalanced output of a CD player, connect the center pin to pin 2 of the balanced XLR input connector, and the ground to pins 1 and 3. To connect the balanced output of something to an unbalanced input requires different techniques depending on whether the output is active balanced (each side has a signal with respect to ground) or floating balanced (for instance, the secondary of a transformer with no center-tap connection). If it's an active balanced output, you can simply use half of it; connect pin 2 to the unbalanced input, and pin 1 to ground, leaving pin 3 floating. If this doesn't work (no or very weak signal) connect pin 3 of the output to pin 1 and ground and leave pin 2 connected to the unbalanced input center pin. Some active balanced outputs, particularly microphones, use the balanced circuit to cancel distortion, so this hookup may result in higher distortion than if a proper balanced-to-unbalanced converter such as a differential stage or a transformer were used.

Q7 - What are ground loops and how can we avoid them?

One of the most difficult troubleshooting tasks for the audio practitioner is finding the source of hum, buzz and other interfering signals in the audio signal. Often these are caused by "ground loops." This unfortunate and inaccurate term (it need not be in the "ground" path, and the "loop" is not what causes the problem) is poorly understood by most users of audio equipment. A better name for this phenomenon is "shared path coupling" because it happens when two signals share the same conductor path and couple to each other as a result.

Another semantic problem that should be addressed early on is the idea that "ground" is one place where all currents go. It's not, there's nothing special about calling a signal "ground," current still flows through any path that's available to it.

Referring to the discussion above regarding unbalanced signal paths, recall that there must be a complete circuit from the output of some device, through the input of another device and back to the "return" side of the output if any current is to flow. Current doesn't flow by itself, it must have a complete path. If there are multiple paths over which the current might flow, the current will be divided among them with most of the current flowing through the path having the least resistance. Any available path, regardless of the resistance in it, will carry some of the current, it's not a case of all the current following the path that has least resistance.

For example, suppose we have two units connected together through a small piece of coaxial cable, and the units are also connected together at the wall outlet through their grounded power cords -- the ground pins are connected to the chassis at each end. The audio signal goes along the center of the coaxial cable, and part of it might come back along the shield of the coax, but part will also go through the ground wire of one unit and back through the ground wire of the other unit. A problem arises when some other signal is also flowing through this same return path. The other signal might be another audio signal, video, data, or power. All of the currents in a wire add together, and the resistance of the wire causes a voltage to appear in proportion to the current flowing. All of these voltages add together, so there is a little bit of the video signal added to the audio, some of the power signal added to the video, some of the power signal added to the audio, etc. In rare instances, the "loop" of wire formed by the intended ground return path and the happenstance lower resistance return path formed by mounting hardware, power cords, etc. can form a magnetic pickup as well, so that magnetic fields radiated by transformers, CRT's, etc. can also induce a current in the "loop," which makes yet another source of noise voltage.

This shared path coupling is a constant problem with unbalanced audio systems. Lots of different methods have been tried to get around the problem, many of them dangerous. Clipping off the ground leads of equipment so there is no common power line path between them simply makes any fault or leakage current follow some other path, back through the signal cable to some equipment that has a ground -- perhaps through the user's body, if all the ground pins have been removed. The only general solution to "ground loop" coupling with unbalanced equipment is to connect all the chassis together with a very low resistance path (copper strap or braid, for example), on the principle that since the resistance is so low, any leakage current will produce a correspondingly low signal voltage. It may also be effective to interrupt the ground path of shield conductors over signal wires; force the return path to go through the designated common strap while leaving the shield in place only for electrostatic screening.

With balanced equipment, no current should be flowing in the shield conductors, and in fact performance should be identical with the shield left disconnected at one end (preferably the receiver end). Therefore balanced systems should be impervious to shared path coupling or "ground loop" problems but in fact they aren't, because most signals inside a given piece of equipment are unbalanced, and there are often return paths internal to the equipment that can be shared with return paths between other units of equipment connected to it. Especially with mixed digital, video and audio signals and high gain, high negative feedback amplifier circuitry, this can be a big problem -- small currents can create big effects -- and this brings us to the next question.

Q8 - What is the "Pin 1 problem" and how do I avoid it?

This is a special case of "ground loop" or shared path coupling. Recently this has been discussed in great detail and clarity by a group led by the consultant Neil Muncy of Toronto. Suppose you have a mixer, whose balanced output is connected to an amplifier's balanced input through a correctly wired cable. Both units are powered from the AC mains and one or both have some small amount of AC leakage current that travels to ground through all available ground paths -- including the shield of the cable that connects the two units. So far so good, no harm done because the circuit is balanced and any common mode voltage from current flowing through the shield will be canceled by the amplifier input. However... a small part of this leakage current also travels through the shield of the wire going from the back panel XLR connector to the PC board, through some "ground" traces on the PC board, and back out through the power line ground cable. No problem so far, except that some gain stage on that same PC board also uses that piece of ground trace in its negative feedback loop, and some part of that leakage signal will be added to the signal in that gain stage; it might be video, or data, or another audio signal, or (most commonly) power.

The solution to this variant of shared path coupling is the same sort of approach that applies to other unbalanced signals: give the leakage current a very low resistance path to follow, and remove as many of the shared paths as possible. Within a unit of equipment, all the XLR connectors' pin 1 terminals should be connected to ground with very low resistance (big) wire or traces, and preferably all of the ground connections should be made at one point, the so-called "star ground" system. A brute force approach is to assume that the back panel is the star ground, and wire every connector's pin 1 solidly to the panel as directly as possible, and lift all the ground wires but one that go from the connectors to the circuitry. In this way, all the external leakage currents (the "fox" to use Neil Muncy's term) will be conducted through the back panel and out of the way, rather than running them through the ground traces on the PC board where they will mix with internal low level signals in high gain stages (the "hen house"). Individual wires can be run from points on the circuit board that need to be at "ground" potential to a common point on the back panel, which is designated a "zero signal reference point" (ZSRP). Equipment that has a reputation for being "quiet" and easy to use in many different applications is often found to be wired this way, while equipment that is "temperamental" if often found to be wired in such a way that leakage currents are easily coupled to internal signal lines. There's a simple test that can be done to check equipment susceptibility to this problem. Connect the output, preferably balanced and floating, of an ordinary audio oscillator to the pin 1 of any two XLR connectors on the equipment. Now operate the equipment through its various modes, gain settings, etc. You may be surprised to find the audio oscillator's signal appearing in many different places in the equipment.