DIRECT INJECTBOX

The present invention relates to a signal processing device, such as a "direct-box" or "direct inject box", for converting a single ended (unbalanced) audio signal from an audio source to a differential (balanced) signal suitable for a microphone input on a mixing desk or similar.

An audio direct inject box (DI box), is used to take a single ended audio signal from an audio source, such as a musical instrument (a guitar, keyboard etc.), or a D J's mixing table, and convert it to a differential, signal suitable for a microphone input on a mixing desk, generally a 600 Ohm signal. The single ended signal connection is typically a 1/4" jack. The differential microphone signal typically uses 3 pin XLR connectors. Most mixing desks have single ended line inputs on 1/4" jacks as well as differential microphone inputs, so that most single ended audio sources could be plugged straight into a mixing desk. However, the mixing desk will often be at a remote location from the audio source. In a recording studio the mixing desk is typically in a separate room, the control room, to that in which the audio sources are located. In a live performance situation, the mixing desk would normally be situated towards the back of a venue, a long way from the performers on stage. It is impractical to run long lengths of single ended, coaxial audio cables between audio sources and a mixing desk, as they will be prone to interference and ground loop mains hum problems. Therefore, the differential microphone inputs of the mixing desk are used because of their hum and interference rejection qualities. In the case of a live performance many triaxial microphone cables will be run between the desk and the stage. These cables usually take the form of a multicore cable for convenience with a stage box at the stage end with, for example, 24 XLR sockets. Microphones can be plugged straight into this stage box using short microphone leads, but single ended, signals from

instruments such as keyboards etc. need to.be converted to differential, microphone level

signals so that they can be plugged into the stage box and consequently be transmitted to the mixing desk. DI boxes are used to convert these signals from single ended to differential microphone level signals

There are generally two types of DI box, passive and active. Passive DI boxes have a simple audio transformer in them to convert the single ended signal to a differential signal.

An active DI box typically has a small buffer amplifier in it and converts the signal to a differential signal using either a transformer, or another inverting amplifier to create the negative half of the differential signal. One advantage of an active DI box over a passive box is that it can have a much higher input impedance thus putting less load onto the instrument connected to it. This can be important for example with inductive guitar pickups, as loading can affect the high frequency response of these devices. Also, an active DI box can have gain, which is useful for weak signals. The active circuitry is quite often powered by power fed down a microphone lead from the mixing desk. This is known as phantom power. The signal level that a DI box receives can vary widely from application to application and whilst the microphone input of a mixing desk will have an input, gain control the input cannot cope with very high signal levels. For this reason most DI boxes will have a selectable attenuator (known as a pad), on the input so that a high signal level can be attenuated. Examples of known active DI boxes are given in US 6,020,788 A and US

2004/0151330 Al.

During set-up for a live performance, before sound checking, the equipment is plugged in on stage by the operator. (Quite often this is a sound engineer). The operator will

then go to the desk (generally at the other end of the venue), to perform a sound check. It is only when the operator reaches the desk that the correct position for the pad switch can be

ascertained and, unless the operator has an assistant, he now has to walk the length of the hall and back to adjust the pad switch. In some cases there is no time for a sound check and after set-up the system is used straight away. In this case if the pad switch is set wrongly then the signal level will either be insufficient, or it will be too high leading to overload of the DI box or desk input and undesirable distortion.

To solve this problem, it is known to provide a signal level meter in a DI box. This allows the signal level to be checked while plugging an instrument into the DI box on stage and for the pad switch to be adjusted accordingly to set a desired level of attenuation.

However, this signal level meter does not address the problem of controlling the DI box output during changing input conditions. The inclusion of a signal level meter only allows the user to see the signal level, and does not provide any additional means of controlling the signal level.

For example, during a live performance a musician may decide to turn up the volume control on his instrument. The result is a change in input signal level, which can lead to degradation of the sound quality at the mixing desk as the attenuation set by the pad switch may be incorrect for the new signal level. Additionally, the musician or the sound engineer may want to change the sound of the instrument. This is possible by using additional devices, such as dedicated attenuation devices for compressing or limiting the signal level. However, adding more devices will increase the number of cables and connections. This is disadvantageous as the set up time is increased and the equipment takes up more space on stage.

Another problem with known devices is that it can be hard to identify faults in the connections to the mixing desk. For example, if one conductor of the microphone lead has a bad connection then half of the phantom power is unavailable, and also only half of the audio signal gets to the mixing desk. Often this fault is not immediately obvious, and it can be hard

to identify which of the two conductors is causing the fault. It would be advantageous if the operator could be quickly alerted of a fault, and where it had occurred.

Viewed from a first aspect, the present invention provides a device comprising an input terminal for receiving an unbalanced electrical input signal from an audio source; means for converting the input signal to a balanced output signal; a variable attenuator for modifying the level of the output signal; processing means for monitoring the level of the input signal and controlling the attenuator based on the level of the input signal; and an output terminal for connection to a mixing desk or similar.

The provision of a variable attenuator, the level of which is set by processing means that measure the level of the input signal, allows the device to adapt to changes in the level of the input signal, so that the quality and level of the output signal can be optimised for any input signal level.

In a preferred embodiment, the processing means comprises a single chip microprocessor. Preferably the processing means comprises a plurality of Analogue to Digital converter channels .

The use of a microprocessor allows the device to be re-programmed. This means that it can be adapted to work with different equipment and instruments, and can be updated with improved software to improve the performance of the device.

The attenuator may be adapted to compress the signal level/to limit the signal level and/or to provide automatic gain control. Preferably a switch is provided to allow a user to switch between the compression, limiting and automatic gain control functions of the attenuator.

This allows the attenuator to control the signal levels appropriately. An automatic gain control allows the device to maintain the .output to the mixing desk at a desired level. By attenuating the signal in different ways the device can adapt to the changing output signal

of the instrument in different ways depending on the user's requirements . For example, unexpected peaks in the signal level can be attenuated by limiting or compressing the signal, in order to maintain sound quality, and avoid damage to equipment. This gives the user greater control over the output signal, without the need for further equipment to be used on stage or at the mixing desk.

In a preferred embodiment a display is provided which is controlled by the processing means for displaying the level of the input signal. Preferably the display comprises a number of LEDs, the number and /or position of the LED or LEDs which are lit in use depending on the detected level of the input signal. In a preferred embodiment the device is connected in use to a mixing desk via a signal conductor, and the device is powered by a current supplied from the mixing desk via the signal conductor. Preferably the device includes means for monitoring the power supply from the signal conductor and producing an error signal if the level of the power supply is incorrect. This is advantageous as it allows any error in the power supply to be detected when the device is being plugged in on stage so that the error can be rectified in the most efficient manner possible. Preferably the signal conductor comprises a microphone line comprising two signal conductors and half of the power supply is transmitted to the device via each of the respective signal conductors.

In a preferred embodiment the means for monitoring the power supply is a part of the processing means.

By using the same processing means to monitor the power supply, to monitor the signal level and to control the attenuator, the device can be constructed efficiently.

The provision of means for monitoring the power supply and producing an error signal is considered to be novel and inventive in its own right and so, from a second aspect, the present invention provides a device comprising: an input terminal for receiving an

unbalanced electrical input signal from an audio source; means for converting the input, signal to a balanced output signal; an output terminal for connection via a signal conductor to a mixing desk or similar such that in use, power is supplied to the device from the mixing desk via the signal conductor; and means for monitoring the power supply in the signal conductor and producing an error signal if the power supply level is incorrect.

This has the advantage that the user is quickly alerted to problems with the power supply.

In a preferred embodiment the signal conductor comprises a microphone line comprising two signal conductors and half of the said power supply is transmitted to the device via each of the respective signal conductors .

The error signal can indicate which of the two signal conductors is at fault, and therefore the user can quickly identify and rectify the fault.

The means for monitoring the power supply may be processing means.

The device of the second aspect of the invention may include any of the features of the device of the first aspect, and also any of the preferred features discussed above.

Preferred embodiments of the present invention will now be described by way of example only and with reference to the accompanying drawings in which;

Figure 1 shows a schematic of a typical use of a direct inject box; and

Figure 2 is a block diagram of an embodiment of a circuit for implementing the invention.

In Figure 1 a direct inject box (DI box) 1 is shown connected to an audio source 2 and a mixing desk 3. The audio source 2 can be a keyboard or other instrument. An on stage amplifier 4 may also be connected to the DI box 1. The audio source 2 is generally connected to an input 11 of the DI box 1 by a 1/4" jack. The output 12 of the DI box 1 is generally a three pin XLR connector, and the output 12 is connected to the mixing desk 3 by

a microphone line. As shown by the figure, the DI box 1, audio source 2 and (if required) amplifier 4 are usually placed at a remote location from the mixing desk 3. Thus, for example in the case of a live performance, the DI box 1, audio source 2 and amplifier 4 are placed on stage, whereas the mixing desk 3 is front of house, usually towards the back of the venue some distance from the stage. In a recording studio, the mixing desk .3 will be in a separate room.

A stage box or similar (not shown) may be used at the stage end, connected between the mixing desk 3 and the DI box 1.

The DI box 1 receives a single ended audio signal from the audio source 2 and converts it to a differential 600 Ohm signal suitable for a microphone input on the mixing desk 3. The DI box 1 is of the active type, as discussed above, and therefore requires a power source. The active circuitry is powered by phantom power supplied from the mixing desk, as a DC voltage on the two signal conductors of the microphone line.

A selectable attenuator (pad) 13 is provided in the DI box, which allows the input signal level to the DI box to be attenuated so that when output from the DI box 1 it is suitable for the mixing desk. Typically this pad could be set to three different levels, OdB (no pad), -2OdB and -4OdB via a switch.

The DI box 1 further comprises a signal level meter 14, which allows the operator to check the level of the input signal directly at the DI box 1 rather than at the mixing desk 3. The signal level can be adjusted using the pad switch during set up of the system for a performance or as part of a sound check. The signal level meter 14 is used to monitor the effect of any adjustments made, either to the pad switch, or to the input itself, for example due to changes in the volume control on the audio source 2.

Additionally, as the signal level meter 14 is powered by the phantom power supplied down the microphone lead, if the level meter 14 works then this indicates that the audio lead

to the mixing desk 3 is intact. Therefore, when the operator gets to the mixing desk he not only knows that the output from the DI box 1 will have the right order of signal level, but he will also know that the various connections and devices are working.

The level meter 14 is driven by a single chip microprocessor that incorporates several A to D converter channels. One of these A to D channels is used to convert the analogue input signal to a digital signal so as to monitor the audio level of the input signal and determine the output of the level meter display 14. The signal level meter display 14 comprises a number of LEDs arranged in a row such that the number of LEDs which are illuminated increases with an increasing level of the input signal, or alternatively such that the position of the lit LED moves along the row with the varying input signal. The number or position of the LED or LEDs which are illuminated at any time therefore depends on the level of the digital signal obtained via the A to D converter. Two other A to D conversion channels in the microprocessor are used to monitor the DC voltages on the two signal conductors of the microphone line. The DC voltages are present because of the phantom power from the mixing desk.

As the microprocessor monitors the level of the input audio signal it can be used to control a variable input attenuator device, such as a symmetrical junction FET provided in the DI box 1. In this way an audio limiter or compressor or an automatic gain control (AGC) is implemented. Thus the audio signal can be modified automatically by the DI box 1, based on the input audio level and a predetermined condition programmed into the microprocessor, such as a maximum signal level.

As the DI box 1 is capable of modifying the audio signal automatically, it can be used to affect the signal levels in various ways. When the attenuator device is used as a compressor, as discussed above, the quiet sounds become louder, and the loud sounds become quieter. Thus the dynamic range of the signal is compressed. With a limiter in

contrast, low signal levels are unaffected, but the signal is prevented from exceeding a set limit by attenuation of high signals. The AGC mode automatically adjusts the gain of the entire signal to prevent the signal levels from going too high. In AGC mode the dynamic range is therefore not affected. A further difference between compressor/limiter mode and AGC mode is in the attack and release time constants, or ballistics of the gain control. The compressor/limiter is generally fast, i.e. it has a short attack and release time. Thus the compressor/limiter quickly changes the gain when a signal is applied, and the gain quickly returns to normal after it has been reduced. The different modes can be selected by a user by means of a switch. Figure 2 is a block diagram of a circuit for the DI box 1. The input signal arrives via the Vi" jack socket 11 from the instrument and is presented to the selectable attenuator 13 via DC blocking capacitor Cl. This selectable attenuator or pad 13 is formed by Rl,2,3 & Sl, and is used by the operator at set up time to adjust the signal level to approximately the correct level as indicated by the bar graph signal level meter 14. After passing through the voltage controlled attenuator (VCA), the signal is buffered by op-amp IClA and inverted by op-amp IClB. Together the outputs of these two op-amps form a buffered, differential signal suitable for feeding to the mixing desk 3. ■

Dl and D2 together with R6 and C4 form a rectifier circuit. This rectifies the signal to obtain a DC signal representative of the audio signal level. This is fed to the A to D converter, which is connected to the microprocessor bus, thus allowing the software access to signal level information.

The software uses this information, firstly to decide which LED(s) to light in the bar graph display 14. The higher the signal, the more LEDs will light or the further the lit LED travels to the right if the graph is used in dot mode. The software does some averaging of the signal level reading to make the attack and decay of the bar graph display 14 more

aesthetically pleasing. It is best that the software rather than the hardware is used to perform these attack and decay calculations because the same raw signal level measurement is used for other purposes, namely limiting and AGC functions, which require different attack and decay rates. Since the microprocessor can control the voltage-controlled attenuator by means of the D to A converter the software can also attenuate the signal automatically if it is too high. Typically a limiter or compressor will increase attenuation as the signal level rises above a certain threshold in order to decrease the amount of signal level rise. The amount of increase of output signal allowed for an increase of input is usually expressed as a ratio. Therefore if the threshold were set to OdBm and the ratio were set to 4: 1 then the following input levels would produce the following output levels :- -8 -8 _4 _4 '

0 0 +4 +1

+8 +2 and so on.

This sort of characteristic is very useful in a live situation where a musician turns up the level of his instrument during the show. Also instruments like acoustic guitars or electric bass guitars can sometimes produce large transients that can be undesirable, giving rise to a rather 'spiky' sound. The limiter is very useful for smoothing out the sound from this kind of instrument. In this instance the attack and decay can be made fairly short so that after the transient has passed, normal gain is restored.

As an example, after suitable amplification a guitar may produce transients of perhaps 5V peak, but yield an average signal level of only 50OmV. This represents a 10: 1 or 1OdB

peak to average ratio. The peaks are produced as the pick strikes or releases the strings, and the average is predominantly the normal decay of the note before the next note is played. By using compression, the same guitar may have the maximum level reduced to perhaps IV. The average level can now be higher as well. Peak to average ratio may be reduced to 6dB or less, and the note will be sustained close to the peak level for much longer than an uncompressed signal.

The limiter/compressor can also be used in other ways to modify the sound of the audio source. For example, many instruments produce a note that decays quickly with time, but it is often desirable to have a long sustained note. As the signal level fades, the compressor can increase its gain, so that the note lasts longer. Compression ratios of perhaps 2: 1 are common, so the output will rise (or fall) by one unit for every two units of input change. A 5OdB dynamic range is therefore reduced to 25dB (from softest to loudest signal). Unlike limiting, the compression threshold is typically set lower than the peak level; the actual threshold level could be anything from +8dB to -4OdB, depending on the effect desired.

Another use is to place a cap on the maximum level of a signal. By using a limiter in this way, any signal below a threshold value is unaffected (although a fixed gain or loss may be applied as appropriate). Above the threshold, signals are attenuated by an amount determined by the compression ratio. A limiter/compressor can also be used in a similar way on other. string instruments, on percussion, on vocals, in fact on any audio source that would benefit.

AGC mode is very similar to the limiting function but the attack and decay times are much slower. An example use for this mode would be where the DI box 1 is used to buffer the output of a DJs console. Pre-recorded music, especially vinyl records, can vary in recorded level quite a lot from record to record. An automatic gain control is useful here to

make all the records come out the same level. DJs are also notorious for turning up the levels as the evening progresses, much to the annoyance of the pub he address engineer, who constantly has to keep an eye on the levels going through his public address system.. Slow attack and decay are preferred in this situation so as to keep the music at a constant level, but 5 not to compress the dynamics of the music.

Thus, the AGC mode does not take any dynamics out of the audio signal, but will just keep the signal levels from going too high. The relative signal levels between the loud and quiet sounds will be unchanged but the whole signal can be attenuated to avoid the peak signal level from exceeding a predetermined threshold. The threshold value would generally

10 be the maximum input level of the mixing desk, or of other equipment using the output signal from the DI box 1. This avoids distortion and overload of the equipment when the input signal level from the audio source 2 increases, as the AGC can automatically adjust the gain . in the DI box 1, so that the signal received by the mixing desk 3 is at an acceptable level. This situation could occur if the musician wants to increase the volume of speakers attached

L 5 to the instrument (audio source 2) via the amplifier 4 and DI box 1. The AGC allows this to occur without the need for the sound engineer to make consequential adjustments to avoid overloading.

Switch S2 is used as a multi function control, in conjunction with the indicator LEDs, to set up the mode and threshold of the limiting or AGC function. LEDl 1 and LED 12 status

10 LEDS are used to indicate limit or AGC mode. Th e bar graph LEDs 14 are used to indicate the threshold level setting while the adjustment is being made. S2 is a three position, centre biased switch. Operating the switch momentarily adjusts the threshold level. Holding the switch for a longer period switches between the two modes.

The microprocessor can also monitor the DC voltages on the two signal conductors of

^5 the microphone line as discussed above. As a result it can be used to detect a fault on these

conductors. When the DI box 1 is powered by phantom power, then half of the power available is supplied by each of the two signal conductors of the microphone line. Therefore, if there was a fault on one of these conductors then only half of the power would be available to power the electronics. Also only half of the differential audio signal would get back to the mixing desk 3. It could however initially appear to the user that everything was working as the level monitor 14 would still function. Therefore, it would be useful for the operator to be alerted to a possible fault. When the microprocessor in the DI box 1 detects that one of the two voltages is wrong, then an indication that there is a fault, and in which conductor it is, is provided by, for example, lighting an LED. In an alternative embodiment the DI box 1 could be battery powered.

Additionally, it is envisioned that the DI box 1 could be made with a signal level meter display 14 that used an alternative to LEDs, for example a liquid crystal display could be used. Alternatively, there could be no display. Changing or removing the level meter display 14 could be used to simplify the display/control panel, or to allow room for other displays. There could also be other means to provide an indication of when the AGC or compressor/limiter is in operation.

The DI box 1 could be adapted so that switching between the various attenuation modes could occur remotely. Thus, a sound engineer at a remote mixing desk could change the mode if necessary without having to go to the stage. This would be useful if there was no opportunity to go back and change the mode on the DI box 1 itself, for example after a performance had started.