GB2488182A - Phantom powered audio dynamics processor - Google Patents

Abstract

The invention relates to a phantom-powered, balanced audio dynamics processor suitable for direct connection to conventional balanced output electrodynamic or capacitor microphones. The device provides dynamic range compression. The device includes a low-noise, low-power balanced analog variable gain amplifier (VGA) based on a long-tailed pair TR1 & TR2, and a microprocessor-controlled side chain for level detection, dynamics processing and linearization of gain control. In an embodiment, the microprocessor IC1 effects control of gain by varying the conductivity of FET TR3. The microprocessor also controls the user interface and display. The VGA and the microprocessor are arranged in a series totem-pole configuration which includes the indicator LEDs.  In embodiments, the device is sufficiently small to fit within the shell of an XLR male to female adapter, a quarter inch jack plug, or within the body of the microphone itself. The form factor of the product and the fact that it requires no separate power supply or batteries, means that the benefits of compression (specifically improved audibility and intelligibility of the human voice) may be exploited where microphones are used (e.g. in public address systems in noisy environments such as railway stations or airports) but compression has hitherto been too inconvenient.

Description

AUDIO DYNAMICS PROCESSOR

FIELD OF THE INVENTiON

The invention relates to devices for dynamic range compression of audio signals, in particular to such devices for dynamic range compression of audio signals from a microphone in e.g. a public address system.

BACKGROUND AND PRIOR ART

Dynamic range compression, also called DRC or simply compression, is a process that reduces the dynamic range of an audio signal, that is, narrows the difference between high and low audio levels or volumes, whilst minimizing introduced distortion. Compression is commonly used during sound recording, live sound reinforcement, and broadcasting.

Compression is applied by running an audio signal through a dedicated electronic hardware unit or through software in audio applications. In the context of audio production, the device is simply referred to as a compressor. In simple terms, a compressor is an automatic volume control. Using downward compression, loud sounds over a certain threshold are reduced in level while quiet sounds remain untreated. Upward compression involves making sounds below the threshold louder while the louder passages remain unchanged. Both reduce the dynamic range of an audio signal. This may be done for aesthetic reasons or to deal with technical limitations of audio equipment, which is seldom able to cope with the dynamic range the human ear can tolerate.

Compression improves audibility of audio in noisy environments, where background noise can overpower quiet sounds so that a comfortable listening level for loud sounds makes quiet sounds inaudible below the noise floor while an audible level for quiet sounds makes loud sounds too loud. If compression reduces the level of the loud sounds but not the quiet sounds the level can be raised to a point where quiet sounds are audible without loud sounds being too loud. A compressor reduces the level of an audio signal if its amplitude exceeds a certain threshold. The amount of gain reduction is determined by ratio: a ratio of 4:1 means that if input level is 4dB over the threshold, the output signal level will be I dB over the threshold. The gain (level) has been reduced by 3dB. Figure I shows the input-output characteristic of a typical compressor.

Compressors are additionally often supplied with attack and release controls that can slow down the response speed of the circuit to smooth the effect and reduce unwanted artifacts such as intermodulation or harmonic distortion.

Commonly available compressors are either mounted in a standard 19" equipment rack, or available in stomp-box pedal format. They require batteries or a dedicated power supply to operate, and must be connected to the rest of the audio system via interconnect cables, which may be balanced (typically XLR or tip-ring-sleeve jacks) or unbalanced (typically tip-sleeve jacks, BNC or RCA phono sockets). They typically operate at line-level rather than microphone-level voltages and impedances, meaning that they are connected into the signal chain after the microphone preamplifier, but before the audio power amplifier stage. This makes them cumbersome for use in many traditional areas where public address systems are employed. Many public address systems do not feature suitable injection points where a known compressor could be inserted between the microphone preamp and the power amplifier.

A TYPICAL MICROPHONE PREAMPLIFIER CIRCUIT

Figure 2 shows an example of a typical phantom powered microphone preamplifier circuit found in public address amplifiers and mixing console microphone inputs. This is the type of circuit to which the dynamic range compression device of the present invention (described below) is intended to be connected. The output of the device is connected to the input of the microphone preamplifier circuit via XLR socket SKI.

Phantom power switch SWI is shown in the ON position; the configuration with which the device is intended to be operated. DC (direct current) is supplied to pins 2 and 3 of the XLR socket SKI via matched resistors Rl and R2. DC blocking capacitors CI and C2 prevent the DC bias voltage from appearing at the inputs of the amplifier IC.

The differential gain of the circuit is designed to be high, whilst the common mode gain and noise should be as close to zero as practical. The current necessary to power the device is drawn through resistors RI and R2 in this circuit. Remaining voltage is passed along to the microphone, so that the device may be operated with microphones which require phantom-power such as traditional condenser (capacitor) type microphones.

SUMMARY OF THE INVENTION

At its most general, the present invention proposes an audio dynamic range compression device that draws its operating power from a phantom power supply of a microphone preamplifier, and/or that is sufficiently small to be entirely enclosed within the housing of a standard audio connector/adaptor or within the housing of a microphone.

In a first aspect the present invention provides an audio dynamic range compression device for compressing the dynamic range of an audio signal sent from a microphone to a microphone preamplifier, the device having a first connector adapted to connect to a microphone and a second connector adapted to connect to a microphone preamplifier, and the device being arranged to draw its operating voltage from a phantom power supply of a microphone preamplifier to which the second connector is connected.

In this way, there is no need for a separate power supply for the compression device.

Prior art compressors rely on batteries, which can make them cumbersome and liable to lose power, or a mains power hook up, which can be inconvenient in some locations. The present invention overcomes these problems by drawing power from the phantom power supply of a microphone preamplifier.

The compression device is connected between a microphone and microphone preamplifier, and thus operates at microphone-level voltages and impedances (typically -6odl3Vrms to -3OdBVrms; 1000 to 6000) rather than line-level voltages and impedances (typically -2OdBVrms to +lOdBVrms; ikO to lOkO). Prior art compressors are connected between the microphone preamplifier and power amplifier, and thus operate at line-level voltages and impedances. Many existing public address systems do not have suitable connection points where a prior art compressor could be inserted, but will have a connection between microphone and microphone preamplifier which can accommodate a device according to the present invention. For example, known microphones have an XLR connector for connection to a microphone preamplifier, and the device of the present invention may be connected via the second connector to that XLR connector.

The device may include a variable gain amplifier and microprocessor arranged in a totem pole' configuration (in series with) whereby current supplied by the second connector passes first through the variable gain amplifier and then through the microprocessor, the microprocessor being arranged to vary the gain of the variable gain amplifier. Thus, the second connector, variable gain amplifier and microprocessor may be arranged in series in that order.

In this way, the variable gain amplifier and microprocessor are powered in series using the same supply current. This is important because of the limited current available from the high voltage, low current phantom voltage supply.

The microprocessor carries at least some of the side-chain, so that the side-chain is largely in software. This helps to minimise the size of the device, and also to minimise its power consumption.

The variable gain amplifier may comprise a long tail pair transistor amplifier.

Additionally, the long tail pair transistor amplifier may comprise first and second transistors, each transistor having an emitter, and the device may include a junction gate field-effect transistor (JFET) connected across the emitters of the first and second transistors.

Alternatively, the long tail pair transistor amplifier may comprise a first resistor connected to a second resistor, and a first transistor and a second transistor, each transistor having an emitter and a collector, wherein the first resistor is connected to the collector of the first transistor and the second resistor is connected to the collector of the second transistor, the device further including a current shunt connected from the connection between the first and second resistors to the emitters of the first and second transistors. Such an arrangement is referred to as a constant current variable transconductance amplifier stage (CCVTAS) in the further

description below.

The device may further include an indicator LED arranged in a totem pole' configuration (in series with) the variable gain amplifier and microprocessor whereby power supply current from the second connector passes first through the variable gain amplifier, then through the microprocessor, and then through the indicator LED, the indicator LED being arranged to indicate the gain reduction of the variable gain amplifier. Thus, the second connector, variable gain amplifier, microprocessor and indicator LED may be arranged in series in that order.

In a second aspect the present invention may provide an audio connector or audio adaptor having a housing, the housing entirely enclosing an audio dynamic range compression device according to the first aspect.

In a third aspect, related to the second aspect, the invention may provide an audio connector or audio adaptor having a housing, the housing entirely enclosing an audio dynamic range compression device for compressing the dynamic range of an audio signal sent from a microphone to a microphone preamplifier, the audio connector or audio adaptor having a first connector adapted to connect the audio dynamic range compression device to a microphone and a second connector adapted to connect the audio dynamic range compression device to a microphone preamplifier.

The second and third aspects enable the device to be easily inserted in an existing public address system, between the microphone and microphone preamplifier. By providing the device within the shell of a standard audio adaptor or connector, it is easily portable and will readily fit into existing systems.

The second or third aspect may comprise an XLR male to female adaptor, wherein the first connector of the device may comprise a male XLR connector and the second connector of the device may comprise a female XLR connector. In this way, the device can be easily connected to an existing microphone female XLR socket. No additional cables or connectors are required, because the cable that originally connected the microphone and microphone preamplifier can simply be plugged into the female XLR connector of the device.

Alternatively, the second or third aspect may comprise a quarter inch jack plug.

In a fourth aspect, the present invention may comprise a microphone having a housing, the housing entirely enclosing an audio dynamic range compression device according to the first aspect.

In a fifth aspect, related to the fourth aspect, the present invention may comprise a microphone having a housing entirely enclosing a microphone circuit and an audio dynamic range compression device for compressing the dynamic range of an audio signal sent from the microphone circuit to an external microphone preamplifier, the microphone having a connector adapted to connect the audio dynamic range compression device to a microphone preamplifier.

The fourth and fifth aspects provide a simple plug-and-play solution for a user, by incorporating the dynamic range compression device of the present invention within a standard microphone housing.

The device provides dynamic range processing (compression, expansion and noise gating) directly in-line between the output of the microphone and the input of a microphone preamplifier which is typically built into a public address amplifier or mixing console. The invention may be connected to the microphone output andfor to the amplifier input. In one embodiment the form factor of the product is an XLR male to female adapter, thus requiring no special cabling or interconnects. Since the product draws its power directly from the preamplifier, it requires no batteries or separate power supply unit. In one embodiment the electronics comprising the device are sufficiently small that they may be embedded within the body of the microphone itself. The device features a simple yet flexible microprocessor controlled user interface with multi-function buttons and LEDs to indicate status.

The optional features of the present invention described above may be applied to any of the aspects of the invention, in any combination.

In summary, the present invention in its various forms provides the following features and advantages: 1. An audio dynamics processor which does not require interconnect cables as its form factor is that of an audio connector or audio adapter.

2. An audio dynamics processor which is entirely contained within the shell of an XLR male to female adapter connector.

3. An audio dynamics processor which is entirely contained within the shell of a standard quarter inch tip-ring-sleeve (TRS) jack plug.

4. An audio dynamics processor which is entirely contained within the body of the microphone.

5. An audio dynamics processor which draws it power source entirely from the phantom power supply of a microphone preamplifier designed with such as supply (such as a preamplifier intended for use with traditional condenser microphones).

6. An audio dynamics processor which features balanced (differential) inputs and outputs which are optimized for low noise operation at microphone levels (typically -6OdBVrms to -3OdBVrms; 1000 to 6000) rather than at line levels (typically -2OdBVrms to +lOdBVrms; ikO to lOkO).

7. A constant current variable transconductance amplifier stage (CCVTAS), which varies differential transconductance over several decades whilst retaining a constant quiescent current.

8. An audio dynamics processor which uses an embedded microcontroller for calibration and linearization of an analog variable gain element such as a junction field effect transistor (JFET), constant current variable transconductance amplifier stage (CCVTAS) or light dependent resistor (LDR), when connected in the feedback path of a differential amplifier such as a long tail pair transistor amplifier.

9. A balanced audio variable gain amplifier where the gain is controlled by a low pass filtered pulse width modulated digital signal driving the gate of a junction FET which is biased into the variable resistance region, connected across the emitters of a long tail differential transistor pair.

10. A balanced audio variable gain amplifier where the gain is controlled by a pulse width modulated digital signal controlling a switched resistor which is connected across the emitters of a long tail differential transistor pair.

11. A balanced audio variable gain amplifier where the gain is controlled by a low pass filtered pulse width modulated digital signal controlling a constant current variable transconductance amplifier stage (CCVTAS).

BRIEF DESCRIPTION OF THE DRAWINGS

Embodiments of the invention will now be described with reference to the appended drawings, in which: Figure 1 is a chart illustrating the input-output characteristics of a typical prior art compressor; Figure 2 is a circuit diagram of a typical prior art phantom powered microphone preamplifier; Figure 3 is a circuit diagram of an audio dynamic range compressor according to a preferred embodiment of the present invention; Figure 4 is a circuit diagram of an audio dynamic range compressor according to an alternative embodiment of the present invention; Figure 5 is a circuit diagram of an audio dynamic range compressor according to another alternative embodiment of the present invention; Figure 6 is a circuit diagram of an audio dynamic range compressor according to another alternative embodiment of the present invention; Figure 7 is a circuit diagram of an audio dynamic range compressor according to another alternative embodiment of the present invention: Figure 8 shows an image of an audio dynamic range compression device according to an embodiment of the present invention; and Figure 9 shows a software block diagram for software suitable for use in an embodiment of the present invention.

Figure 10 is a hardware block diagram showing the arrangement of subsystems in a lotem pole' configuration.

DESCRIPTION OF THE EMBODIMENTS

In the description of each of the illustrated embodiments like reference symbols are used to describe like circuit elements.

PREFERRED EMBODiMENT (EMBODIMENT A) Figure 3 shows the preferred embodiment (embodiment A) of the circuit diagram of an audio dynamic range compressor according to the present invention. An XLR balanced microphone (not shown) connects to input socket SKi. A microphone preamplifier (not shown) with a phantom power supply connects to output socket SK2.

DC biasing and phantom supply current is supplied from the preamplifier to the microphone via resistors R3, R4, R5 and R6 and diode Dl. Resistors R5 and R6 also provide base voltage biasing for transistors TRI and TR2. Diode Dl provides sufficient VCE to transistors TR1 and TR2 to ensure that the transistors are not in saturation, and have sufficient headroom for the maximum anticipated signal voltage.

Resistors R3 and R4 are equal in value, and in addition to passing DC biasing to the microphone, determine the (series mode) collector impedance and thus maximum differential (series mode) gain of transistor pair TR1 and TR2.

Transistors TRI and TR2 form a long tail pair differential amplifier, with tail current determined by current source 12. In the absence of JFET TR3 the gain is approximately determined by R3, R4, Ri and R2. Current source 12 provides improved common mode rejection ratio (CMRR), as the DC voltage on the bases of the long tail pair will vary with downstream microphone current consumption.

JFET (junction field-effect transistor) TR3 is configured as a variable resistance connected across the emitters of TR1 and TR2, which can vary the gain of the long tail pair according to its V applied bias voltage. Since the drain and source of TR3 are at the same potential, VD5 = 0 so the resistance of the drain-source channel is controlled by VGS alone. TR3 is an N channel junction FET (field-effect transistor), which typically has a negative V05 bias voltage applied since it is a depletion mode device.

Shunt D2 provides the power supply voltage for microprocessor IC1, which floats just below the common mode voltage at which the microphone operates. Capacitor C5 provides power supply bypassing for IC1.

LED1 and LED2 provide operating mode and gain reduction indication to the user, and are driven from microprocessor outputs OPI and 0P2. MOSFETs (metal oxide semiconductor field-effect transistors) TR4, TR5, TR8 and TR9 provide logic level shifting to permit the microprocessor to switch LEDI and LED2 whilst its outputs remain within the range of its power supplies. Note that these LEDs may be pulse width modulated (PWM) by the control software running on microprocessor IC1 to vary the brightness of the LED5, for example to indicate the current incoming signal level or the amount of gain reduction being applied.

The device draws its power from the phantom power supply of the microphone preamplifier. Phantom power is a high-voltage, low current supply, and in order to use this effectively the subsystems of the compressor are arranged in a totem pole' configuration (i.e. in series), with the same supply current reused in each. This is illustrated in Figure 10. On top is the variable-gain amplifier formed by the long tail pair TR1 and TR2. Underneath that is the microprocessor IC1. Below that are the indicator LEDs LEDI and LED2.

The variable gain amplifier is a long tail pair run at quite a high current. This is to keep the noise down and so that it can drive a low impedance load. The gain is varied by changing the resistance between the emitters of the transistors of the long tail pair. The JFET TR3 provides this variable resistance. The PWM (pulse width modulated) output from the microprocessor lCl is level-shifted by the level shifter 1C2 and passed through a two-pole low pass filter to get the control voltage for the JFET TR3.

The same current that powered the variable-gain amplifier is reused to power the microprocessor IC1, apart from a small current powering the level shifter 1C2, which of necessity must straddle the voltage range of the microprocessor and the variable gain amplifier. The microprocessor lCl implements the side chain of the compressor.

Its audio input is AC coupled with capacitors C7 and C8 as the microphone input is more positive than the microprocessor.

The same current is used again to power the indicator LED5. This makes their control more complex. A level shifter for each LED is necessary, comprising (TR4, R14 and TR8) and (TR5, Ri 5 and TR9). The overall tail current through all parts of the device is defined by the current sink 12.

By arranging the long tail pair, microprocessor and display LEDs in series in this configuration ensures the minimum possible current consumption from the microphone preamp, whose current supply is typically limited to a maximum of approximately l2mA, but whose common mode voltage is typically around 48VDC with respect to ground (pin I of the XLR connector in this example). The total current consumption from the circuit as shown is approximately 4mA. The current through current sink 12 must be greater than the maximum power supply current of IC1 to enable shunt D2 to operate effectively and maintain regulation of the power supply for microprocessor IC1.

Microprocessor 1C1 controls the gain of the amplifier via a pulse width modulated output PWM. The PWM control signal is level shifted by 1C2 before driving the gate of TR3. Current source II, reference D4 and capacitor C6 provide a suitable power supply for 1C2. Low pass filter network R12, C3, Rh and C2 transform the pulse width modulated control signal into a DC bias voltage which varies TR3 VGS between OV and VGS_OFF, ensuring that the JFET can be turned fully on or off.

Since TR3 appears as a resistance between the emitters of TRI and TR2, as its channel resistance is decreased, the gain of the amplifier is increased. The maximum gain the amplifier can achieve is limited by the minimum RDS_0F4 of TR3, the intrinsic emitter resistance of TRI and TR2 and the effective collector load of TR1 and TR2. The minimum gain the amplifier can achieve is determined by R3, R4, Rh and R2, and the effective collector load of TR1 and TR2. The difference between these gains is the effective dynamic range of the dynamics processor circuit. With the circuit as shown the compressor has greater than 40dB of gain variation available, which is more than sufficient for most audio compression and expansion purposes.

Switches SWI and SW2 are the user controls for the product, and are connect to microprocessor inputs IP1 and lP2, which have internal pull-up resistors enabled.

The input signal from the microphone is connected via DC blocking capacitors C7 and C8 to a differential analogue to digital (ND) converter within microprocessor IC1.

The input level detection, gain computation, linearization and dynamics processing (attack, decay, sustain, ratio, peak hold, threshold) are computed by the software running in microprocessor lCl, and result in a suitable pulse width modulation signal being generated.

ALTERNATIVE EMBODIMENT B

Figure 4 shows alternate embodiment B. Transistors TRI and TR2 are connected in a long tail pair configuration without emitter degeneration resistors. In this configuration, the transconductance of TR1 and TR2 is a function of their quiescent collector current according to the Ebers-Moll bipolar transistor model. As the collector current is increased in TR1 and TR2, the differential gain of the amplifier wilt also increase due to the increase in the transconductance of the long tail pair. The gain of this circuit configuration is thus controlled by varying the quiescent current through transistors TRI and TR2. This is typically how the gain is controlled in an operational transconductance amplifier (OTA). However in this circuit there must always be sufficient current flowing through current sink 12 for the microprocessor 101 to operate and for its power supply rails to be in regulation, so the total current flowing into pin VCC of IC1 must remain approximately constant.

Current sink 12 provides a constant current sufficient to operate microprocessor IC1, and provides the maximum collector current for TR1 and TR2. Transistor TR3 operates as a variable current source which diverts or steers some of the current flowing through 12 away from the long tail pair TRI and TR2. As the current through TR3 is increased, the collector current through TR1 and TR2 is decreased and the gain of the amplifier is reduced. With TR3 fully turned off, the gain is at a maximum.

With TR3 fully turned on, no collector current flows through TRI and TR2 and the gain is at a minimum.

Pulse width modulated control output from microprocessor lCl output PWM is level shifted by TR6 and filtered by the action of R8 and 02 to provide a DC bias voltage, which in turn controls the current through transistor TR3 via biasing network R7, R2 and Ri. We will hereafter refer to this circuit configuration as a constant current variable transconductance amplifier stage (CCVTAS).

The alternative embodiment of Fig. 4 thus has the same general totem-pole' arrangement of subsystems as the preferred embodiment of Fig. 3, but in this embodiment the variable gain amplifier is implemented in a different way. The long tail pair at the input has no emitter degeneration resistors and so naturally has a high gain. The tail current of the entire block must remain constant, as that current is reused to power other blocks. The tail current of the long-tail pair is varied by bypassing some of the fixed current through the shunt TR3. The gain of the input pair reduces as its tail current reduces.

ALTERNATIVE EMBODIMENT C

Figure 5 shows alternate embodiment C. In this embodiment, light dependent resistor LDRI takes the place of the junction FET in the preferred embodiment (Fig. 3). LDR1 is optically coupled to LED3, such that as the brightness of LED3 varies the resistance of LDRI will vary accordingly, and thereby control the gain of the circuit.

Microprocessor lCl pulse width modulated output PWM drives the brightness of LED3 by varying the pulse width. Note that the current to power LED3 is the same current used to power microprocessor IC1, albeit at a lower voltage. Level shifting occurs via TR3 and TR6 so that microprocessor ICI output PWM remains within its power supply extremes. This configuration is necessary to minimize the current drawn from the phantom powered microphone preamplifier since LED3 consumes a significant amount of current when at maximum brightness.

The alternative embodiment of Fig. 5 has the same overall totem-pole' structure as the embodiments shown in Figs. 3 and 4, as illustrated in Figure 10. However, in this embodiment the gain is changed with an LDR across the emitters of the input pair.

The LDR is illuminated with a LED. In order to get sufficient drive current, this has to be arranged below the microprocessor IC1, in the same way as the indicator LEDs are driven.

ALTERNATIVE EMBODIMENT D

Figure 6 shows alternate embodiment D. In this circuit, the pulse width modulated output PWM from microprocessor ICI is connected to a digitally controlled analogue switch 1C2. Ii, 03 and C2 provide power supply for 1C2 to ensure that its analog switch terminals remain within its power supply voltage range.

Provided that the switching frequency of PWM is sufficiently fast then switch lC2 in series with resistor R7 will appear at frequencies below the Nyquist frequency to be a resistor, whose resistance is inversely proportional to the mark-space ratio of the PWM control signal. The mark-space ratio of PWM will therefore control the gain at audio frequencies. There will be a series of unwanted high frequency signals superimposed on the output (collectors) of the long tail pair in this configuration, as will be familiar to those skilled in the art of sampled signals. Capacitor Ci, in conjunction with TRI and TR2, acts as a reconstruction filter to remove unwanted switching harmonics at the output of the amplifier. For audio purposes the maximum frequency of interest is usually 20kHz. If 1C2 were to be switched at 2MHz, with the pole introduced by Ci set to around 20kHz, then unwanted switching harmonics would be attenuated by approximately 40dB. Further high frequency attenuation could be achieved by additional low pass filtering on the output of the circuit if desired. Most microphone preamplifiers feature some low pass filtering on their inputs which further improves switching harmonics rejection.

The alternative embodiment of Fig. 6 has the same overall structure as that of Figs. 3-5, but the variable resistance between the emitters of the input pair is provided by R7 and the analogue switch 1C2. The PWM driving lC2 is very much faster than the highest audio frequency, so the effective resistance at audio frequencies is controlled by the PWM ratio. This approach puts large amounts of the switching frequency on the output. It is filtered out with Ci.

ALTERNATIVE EMBODiMENT F Figure 7 shows alternate embodiment E. In this configuration, microprocessor ICI is ground referenced, i.e. its power supply is referred to ground. Since the long tail pair TRI and TR2 is floating at a voltage which is dependent upon the phantom power supply voltage of the microphone preamplifier and the current which is drawn downstream by any connected microphone, transistor TR6 is used to level shift pulse width modulated signal PWM up to the potential at which the long tail pair operates.

In this configuration, JFET TR3 is connected across the emitters of a degenerated long tail pair. Resistors Ri 1 and Ri 2 ensure that the TR3 is turned on (Vs = 0) when no PWM signal is present. The PWM signal from microprocessor ICI is smoothed to a DC voltage by Ri 0 and Ci. This voltage controls the current flowing through TR6 which is emitter degenerated to act as a variable current sink.

As the current through TR6 increases, this decreases the V0 of TR3, progressively turning off TR3. Once TR3 has completely turned off, the gain from the amplifier is at a minimum. This configuration has a dynamic range of approximately 40dB, but is considered inferior to preferred embodiment A, primarily due to the Early effect' on TR6 as the downstream current changes, and to temperature sensitivity of the current sink, due to TR6's relationship between VCE, collector current and temperature.

In Fig. 7 the microprocessor ICI is ground referenced, and the control voltage to 1R3 is moved over a considerable voltage span (about 25-30V) by converting it to a current in TR6 and then converting that current back to a voltage, but referenced to the emitters of the input pair, in RI I and R12. It has a totem pole' structure similar to that illustrated in Figure 10, but in this embodiment the order of the subsystems is varied.

SUMMARY OF EMBODIMENTS

Each of embodiments A to E described above has in common that each of the subsystems are arranged in a totem pole' configuration so that the variable gain amplifier is at the top, followed by the microprocessor (Id), and the indicator LED block at the bottom. This totem pole' configuration is illustrated in the hardware block diagram of Fig. 10.

As shown in Fig. 10, the output connector (SK2), variable gain amplifier, microprocessor (Id), indicator LEDs (LEDI, LED2), and current sink (12) are arranged in series so that the variable gain amplifier, microprocessor and indicator LED subsystems each draw an equal current. Arranging the variable gain amplifier, microprocessor and display LEDs in series in this configuration ensures the minimum possible current consumption from the microphone preamp, whose current supply is typically limited to a maximum of approximately l2mA, but whose common mode voltage is typically around 4BVDC with respect to ground.

FORM FACTOR AND CONSTRUCTION

Figure 8 shows one example of how the printed circuit board (PCB) assembly might be attached to a connector shell; many others are possible. Note that the LEDs and buttons may be mounted on to the PCB and mechanically connect to the shell via plastic injection mouldings or suchlike. In this example, one end of the connector is an XLR male connector (output) for connection to the input of a microphone preamplifier. The other end of the connector is an XLR female connector (input) for connection to the output of a microphone which has an XLR connector. One of the XLR connectors is directly soldered to the printed circuit board for stability. The other XLR connector is soldered to the printed circuit board via flying leads to facilitate assembly of the printed circuit board assembly into the connector shell.

SOFTWARE BLOCK DIAGRAM

Figure 9 shows the block diagram of the software running on the embedded microprocessor within the unit. The inputs to the software are the analog signal from the microphone and the state of the buttons from the user interface. The outputs from the software are the pulse width modulated gain control signal and the LED display drive (which may also be pulse width modulated for variable brightness).

Note that in the example shown the software does not measure the analog output signal. The block diagram shown is for a feed-forward compressor. Alternatively one could use an additional analog input for the output signal, and design a feedback compressor with similar performance characteristics if desired.

SOFTWARE BLOCK FUNCTIONS

The following description of the software block functions performed by the microprocessor ICI refers to Figure 9.

* The AID conversion block samples the incoming signal from the microphone and feeds this into the input level detection block.

* The input level detection block full wave rectifies the incoming samples, and features an envelope detector to determine the mean or RMS energy level present in the incoming signal.

* The dynamics processor determines the attack, decay, sustain and compression ratio. Based on the current and historical incoming signal levels it determines what output gain of the variable gain amplifier is required.

* The output calibration block converts a desired gain into a pulse width modulation parameter. Analog elements such as a junction FET, optically coupled light dependent resistor or constant current variable transconductance amplifier stage (CCVTAS) require linearization since the relationship between the PWM mark-space ratio and amplifier gain may be highly non-linear, and may also depend on temperature, input signal level, phantom power supply voltage and so on. The output calibration block performs this linearization.

* The pulse width modulation block drives the output circuit PWM pins with the appropriate mark-space ratio for the desired output gain.

* The LED driver block controls the brightness of the LEDs on the user interface.

* The button handler controls the mode of operation of the unit.

* Parameter storage is where the current settings are contained, for example threshold, ratio, attack time, release time, gate threshold, and gate attack time.

* Calibration storage is typically written during initial programming and test of the device and provides the output calibration block with sufficient information to be able to accurately set an output gain.

Claims (13)

1. CLAIMS1. An audio dynamic range compression device for compressing the dynamic range of an audio signal sent from a microphone to a microphone preamplifier, the device having a first connector adapted to connect to a microphone and a second connector adapted to connect to a microphone preamplifier, and the device being arranged to draw its operating power from the phantom power supply of a microphone preamplifier to which the second connector is connected.
2. 2. A device according to claim 1, including a variable gain amplifier and microprocessor arranged in a totem pole' configuration whereby power supply current from the second connector passes first through the variable gain amplifier and then through the microprocessor, the microprocessor being arranged to vary the gain of the variable gain amplifier.
3. 3. A device according to claim 2, wherein the variable gain amplifier comprises a long tail pair transistor amplifier.
4. 4. A device according to claim 3, wherein the long tail pair transistor amplifier comprises first and second transistors, each transistor having an emitter, and the device includes a junction field-effect transistor (JFET) connected across the emitters of the first and second transistors.
5. 5. A device according to claim 3, wherein the long tail pair transistor amplifier comprises a first resistor connected to a second resistor, and a first transistor and a second transistor, each transistor having an emitter and a collector, wherein the first resistor is connected to the collector of the first transistor and the second resistor is connected to the collector of the second transistor, the device further including a current shunt connected from the connection between the first and second resistors to the emitters of the first and second transistors.
6. 6. A device according to any of claims 2 to 5, including an indicator LED arranged in a totem pole' configuration with the variable gain amplifier and microprocessor whereby power supply current from the second connector passes first through the variable gain amplifier, then through the microprocessor and then through the indicator LED, the indicator LED being arranged to indicate the gain reduction of the variable gain amplifier.
7. 7. An audio connector or audio adaptor having a housing, the housing entirely enclosing an audio dynamic range compression device according to any of claims 1 to 6.
8. 8. An audio connector or audio adaptor having a housing, the housing entirely enclosing an audio dynamic range compression device for compressing the dynamic range of an audio signal sent from a microphone to a microphone preamplifier1 the audio connector or audio adaptor having a first connector adapted to connect the audio dynamic range compression device to a microphone and a second connector adapted to connect the audio dynamic range compression device to a microphone preamplifier
9. 9. An audio connector or audio adaptor according to claim 7 or claim 8, comprising an XLR male to female adaptor, wherein the first connector of the device comprises a male XLR connector and the second connector of the device comprises a female XLR connector
10. 10. An audio connector or audio adaptor according to claim 7 or claim 8, comprising a quarter inch jack plug.
11. 11. A microphone having a housing, the housing entirely enclosing an audio dynamic range compression device according to any of claims 1 to 6.
12. 12. A microphone having a housing entirely enclosing a microphone circuit and an audio dynamic range compression device for compressing the dynamic range of an audio signal sent from the microphone circuit to an external microphone preamplifier, the microphone having a connector adapted to connect the audio dynamic range compression device to a microphone preamplifier.
13. 13. An audio dynamic range compression device substantially as herein described, with reference to the accompanying drawings.Amended claims have been filed as follows:-CLAIMS1. An audio dynamic range compression device for compressing the dynamic range of an audio signal sent from a microphone to a microphone preamplifier, the device a) having a first connector adapted to connect to a microphone and a second connector adapted to connect to a microphone preamplifier b) being arranged to draw its operating power from the phantom power supply of a microphone preamplifier to which the second connector is connected c) containing a microprocessor whose purpose is to sample the incoming waveform and determine the instantaneous gain to be applied d) featuring an analog, balanced, low-noise, differential, variable gain amplifier whose gain is determined by signal supplied by a microprocessor e) including a variable gain amplifier and microprocessor LIt) arranged in a totem pole' configuration whereby power supply 0 current from the second connector passes first through the (0 variable gain amplifier and then through the microprocessor r 2. A device according to claim 1, wherein the variable gain amplifier comprises a long tail pair transistor amplifier.3. A device according to claim 1, wherein the long tail pair transistor amplifier comprises first and second transistors, each transistor having an emitter, and the device includes a junction field-effect transistor (JFET) connected across the emitters of the first and second transistors.4. A device according to claim 1, wherein the long tail pair transistor amplifier comprises a first resistor connected to a second resistor, and a first transistor and a second transistor, each transistor having an emitter and a collector, wherein the first resistor is connected to the collector of the first transistor and the second resistor is connected to the collector of the second transistor, the device further including a current shunt connected from the connection between the first and second resistors to the emitters of the first and second transistors.5. A device according to any of claims 2 to 5, including an indicator LED arranged in a totem pole' configuration with the variable gain amplifier and microprocessor whereby power supply current from the second connector passes first through the variable gain amplifier, then through the microprocessor and then through the indicator LED, the indicator LED being arranged to indicate the gain reduction of the variable gain amplifier.6. An audio dynamic range compression device substantially as herein described, with reference to the accompanying drawings c\J rLU (0 r