High Quality Sound Mixer

High Quality Sound Mixer

http://sound.westhost.com/project30.htm[11.01.2010 19:12:03]

Elliott Sound Products Project 30

High Quality Audio MixerRod Elliott (ESP)

Introduction

This project is probably the most ambitious so far, and can be expected to be very expensive. On thepositive side, it is also capable of excellent performance, and can be tailored to suit your exactspecifications. There are several different input modules in the series, the first being the microphone and lineinput.

The project is presented in parts, and Part 1 shows the mic/line module and has some backgroundinformation on noise measurements and the general philosophy behind the project.

Stages 2 and 3 are now complete, and are described below, and Stage 4 is available, but still underconstruction.

Description

Since the project is presented in stages, this is an index for the various modules, and as the system isdeveloped will also be the place to look for updates and other information.

UpdatedStage 1 Microphone / Line input module - Includes optional 48V phantom feed, and shows

three different input stage configurations. You can choose from transformer input, ortwo different electronically balanced circuits. Also shows the tone control circuits,peak level indicator and all channel to group/master switching, faders and pan pots.

07 Jan2001

Stage 2 Basic Mixing Modules - These are used for mixing the stereo sends from each of theMic/Line modules, either as sub-mixers or the main mixer (the same unit is used foreach). Also included are the Auxiliary Mix Module and balanced line driver circuits,and the first stage of the power supplies.

23 Oct1999

Stage 3 Power Stages - This section shows the power supply regulators - both +/-15V mainsupplies and 48V phantom supply, and the headphone power amps.

26 Nov1999

Stage 4 Bits and Pieces - The next installment will describe the pre-fade listen and otherheadphone mixing and switching, as well as the talkback mic amp, phono andauxiliary input modules. (See NOTE below.) Stage 4 is under construction, andcontains descriptive text only at this time.

The additional modules, metering and a complete system layout will be added to the list as they aredeveloped.

High Quality Sound Mixer

http://sound.westhost.com/project30.htm[11.01.2010 19:12:03]

NOTE: There has been a surprising amount of interest in this project, with a common requirement being asmaller version (an almost equally common request has been for a bigger version, too). Scaling the projectup is not really a problem, but it is difficult to know what you can leave out to make a small mixer of 6channels or less.

It seems I have also managed to confuse a few people with some of the links between the various stages. Iused two different terms for the same one in one instance, and the links made sense to me at the time, butthis has not helped some of my readers. Sorry about this, and I will try to find and fix the errors (whether ofnomenclature or common sense).

There is also some concern about the time taken for Stage 4. I have been busily doing other things, andmanaged to push the mixer to the "back burner" - I am almost sorry I started this, but I will have to continue -after all, I got this far. Bear with me, gentle reader - it won't happen overnight, but it will happen. (After all, Ido have a "real" job - one has to eat ... )

Projects Index

Main Index

Copyright Notice. This article, including but not limited to all text and diagrams, is the intellectual property of Rod Elliott, and isCopyright (c) 1999. Reproduction or re-publication by any means whatsoever, whether electronic, mechanical or electro-mechanical, is strictly prohibited under International Copyright laws. The author (Rod Elliott) grants the reader the right to use thisinformation for personal use only, and further allows that one (1) copy may be made for reference while constructing the project.Commercial use is prohibited without express written authorisation from Rod Elliott.

Project created 22 Sept 1999 - Part 1 added at the same time. Part 2 added 23 Oct 1999. Part 3 added 26 Nov 1999

http://sound.westhost.com/projects.htm

http://sound.westhost.com/index.html

Audio Mixing Console - Part 1

http://sound.westhost.com/project30a.htm[11.01.2010 19:12:53]

Elliott Sound Products Project 30a

High Quality Audio Mixer - Stage 1Rod Elliott (ESP)

Introduction

This is part one of probably the most ambitious project so far, in that it can be huge (there isno real reason that it could not be built up as a 36-8-2 (i.e. 36 input channels, 4 "sub" stereooutput channels and a main stereo output, not including auxiliary sends). The picture on theleft is a representation of how a single mic / line channel might look, with the sub-master (orgroup) switching included. There are modules planned for every input type imaginable,including:

Standard mic / line module with 48V phantom power (electronically balanced or usingtransformers)Stereo phono module with RIAA equalisationStereo Line moduleTalkback mic / headphone amp module

All input modules have 3-band EQ, with the mid frequency variable from 500Hz to 2kHz toallow maximum flexibility. Inputs all have variable gain, and each input module has a peaklevel LED, PFL (Pre-Fade Listen) button, channel insert and auxiliary sends. Optionally (for afour output mixer), there is a simple selector switch to select the master bus (A or B), and allchannels have a pan control for stereo positioning. Stereo input modules have a balancecontrol instead.

The maximum gain of the mic / line unit has been set at 46dB, which is more than enough formost music recording. This allows for microphone levels down to 5mV, which is almostalways exceeded by most musical instruments with a typical low impedance microphone. Inmany cases (especially with vocalists), the mic output level can easily reach 250mV, and Ihave measured the output of a low impedance mic at about 1 Volt with loud singers! Whereextra gain is needed, this is easily accommodated.

Each of the input preamplifier units can easily be changed to give more gain, but noise mustbe considered the greatest enemy of any recording or live music mixer. As gain is increased,noise increases as well.

The nominal operating level of all modules is 0dBv (1 Volt RMS), and the peak LED willoperate at +6dBv. This allows 15dB headroom, which is more than adequate provided themixer is not operated with all the peak LEDs continuously on!

Please note that although presented in project form, this is not necessarily a "real" project.Rather, it is a gathering of ideas with a common theme, and circuits for audio mixing consolesare few and far between on the web from what I have seen.



The Project Continues . . .

The output module is also configurable, allowing a selection of options. As can be imagined,this project will take some time to complete, so will be presented in stages. This is the first ina series, and provides all the information for building the microphone / line input module.

Following is the master module, and this way you can start to build a complete unit, and addthe other modules later. The next is the power supplies, which has much higher currentcapacity than one for an audio preamp and includes the optional 48 Volt phantom feedsupply. This does not require particularly high current, but needs to be very well smoothed, toprevent hum from being introduced into the low level microphone inputs. The phantom feedsupply is designed to handle a maximum of about 10 microphones (or direct injection boxes)at any one time.

Block Diagram of Mic/Line Stage

Figure 1 shows the block diagram of the mic / line input stage with a single stereo output.This is the option most likely to be built, as it is simple to use and is relatively cheap - but onlycompared to more esoteric options. This is not a cheap project to build, but will provide a

standard of performance that is very hard to beat with equivalent commercial offerings.

Although I have designed this mixer with two auxiliary sends, there is no real reason that more could not beprovided. Even with the two pots, a switch can be used to select one of several buses for each control.Generally, the ability to select either of two buses for each auxiliary send is enough, but more can be used ifdesired.

I will leave it up to the individual constructor to determine the ideal combination. Remember that for eachbus, there must be a mixing module (these can be as simple or as complex as you like), and the necessaryoutput connectors for each. This all starts to add up (rather quickly, too), so you do need to consider the finalcost.

Figure 1 - Mic / Line Module Block Diagram

Each mic/line channel has 7 pots, up to 5 switches, one push-button and a slide pot for the fader. In addition,there is a Cannon XLR input connector, a stereo jack for channel insert and the overload LED. Naturally,there is also the electronics to make it all work. An ambitious project indeed, but one that I hope will bepopular despite all of this.



The mic/line input module is the most important part of the system, since it is this unit that determines thefunctionality of the entire mixer. Accordingly, this provides input selection, gain control, phantom power,phase reversal, tone controls, auxiliary sends, pre-fade listen, pan-pot and overload indication. (If you saythat really quickly, it still sounds like a lot of stuff!)

Mic / Line Input Section

This is common to all of the following amplifier units, and provides the following functionality:

Standard Cannon XLR female input connectorPhantom power switching (optional 48V feed for powered microphones and DI boxes)20dB pad for high level line inputsPhase reversal switch for correcting out of phase microphone or line inputsDC blocking capacitors (can be omitted if phantom powering is not used)Protection zener diodes (can be omitted if phantom powering is not used, but I recommend they beused anyway)

Figure 2 - Mic / Line Inputs And Switching

The 48 Volt phantom feed can be omitted if you are quite sure that you will never need it, but it is needed ifpowered microphones are contemplated, and can also be used to power direct injection (DI) boxes so thatbatteries do not have to be used.

The 20dB pad is needed if high level line inputs are going to be used. All of the amplifier options have aminimum gain of 3 (close enough to 10dB), so if a line input of 0dBv (i.e. 1 Volt) is applied the gain controlwill be set near its minimum gain setting. For higher levels, the pad reduces the input signal, allowing up to+20dBv (10V RMS) to be applied without overload. It has always been something of a convention to use a20dB pad, but in my experience these are a bit of a pain. The pad tends to reduce (or increase) the signal byjust that little bit too much, so I thought briefly about a 10dB pad instead. Feel free to modify the circuit if youprefer a 10dB pad.

The phase reversal switch is used where two microphones are used in close proximity, and cause phasecancellation because of their relative distance from the source. This can result in a "hollow" and often



unpleasant sound - especially when miking a drum kit or piano. The phase switch allows you to select thebest setting to get the sound you want.

The capacitors and zener diodes protect the following amplifier from transients when a mic is plugged in orremoved with phantom power applied. These are recommended even if the phantom feed option is not used,especially with the electronically balanced circuits. They are not essential with the transformer input, but willdo no harm.

Click on the PCB image to see Project 96 - a different version of the phantom feed and distribution (and a powersupply) - PCBs are available.

Mic / Line Input Module (Transformer Input)

There are three options for the input, using a transformer to balance the input, or using either of twoelectronic balancing circuits. My preference is (and always has been) for the transformer, as the commonmode performance is far better than the electronic method, and also provides much better radio frequencyinterference suppression. However, transformers are expensive, so both methods are shown, but the noiseperformance will suffer and interference will also be worse if the transformer is not used.

In reality, these differences may not be an issue, and the use of a cheap ($20 - $50) transformer may (will?)degrade performance far more than the use of an electronically balanced circuit.

Figure 3 - Transformer Input Mic/Line Input Circuit

The suggested transformer is a Jensen JT-16-A. This is a 1:2 step up transformer, which provides a usefulgain of 6dB, and more closely matches the impedance to obtain the optimum noise figure from the inputamplifier (an NE5534A single opamp). It is more than probable that most constructors will be stopped by thetransformer (it stopped me, so I did most of my preliminary testing using a 1:1 transformer that I had tohand), but if you can get them this is the best option. Be warned, good transformers are expensive, so if youcan't afford good ones use the next circuit instead. Cheap transformers will degrade the sound to anunacceptable degree, and should be avoided.

Also note that as shown, this circuit has a maximum gain of 40dB. To achieve the 46dB gain mentionedabove, change the 50k pot to 100k.

Mic / Line Input Module (Electronically Balanced Input - Version 1)

The electronically balanced input stage needs some fairly radical protection from the switching transientsproduced when the 48 Volt phantom supply is turned on or off. This is most easily accomplished using the

http://sound.westhost.com/project96.htm



zener diodes as shown in Figure 2. The input circuit is a modified version of what is commonly called an"instrumentation" amplifier, and provides far better impedance balance than the simpler version commonlyused. The impedance balance is very important in this application, because if the impedances are not equalfor the inverting and non-inverting inputs, the noise rejection suffers badly.

The gain is set using the 10k pot VR1. The capacitor C4 helps prevent the circuit from oscillating (theseopamps have a very wide bandwidth), and also act as an RF stopper, along with the two 1k resistors. Thiscombination gives a worst case upper -3dB frequency of about 25kHz, and from tests I conducted isextremely effective.

Figure 4 - Electronically Balanced Mic / Line Input Module

I would rather have liked to have used a "bootstrapped" input circuit, but it is patented (by someone else), sowas not used in this design. The alternative shown is only marginally worse than the bootstrapped design,and should give a fairly good account of itself, even under adverse noise conditions.

There is one major benefit of the arrangement shown, in that it behaves just like a transformer forunbalanced inputs (well, almost exactly). If the source is connected to only one input, the output voltage isnegligible, and it requires that the unused input connection is grounded. Apart from anything else, thisindicates that the external noise contribution is far lower than would be achieved with the simple balancedinput circuit (U3) alone. The input stage shown is variable gain, without any fuss or difficulty.

This circuit has much better impedance balance than the circuits you normally see, because the centre tap ofthe two input resistors is not connected directly to ground. This allows the input circuit to "float" aboveground, and improves common mode noise rejection for inputs that are not perfectly balanced (which is mostof them, due to imperfect leads etc.).

The balancing shown is so good that it is necessary to disconnect the phantom supply completely, or it willdegrade the performance for noise rejection quite badly. The same applies to the transformer input, by theway. In tests I conducted, the unbalanced signal rejection was better than 30dB. This is not magnificent, butit is much better than the 6dB from the conventional single opamp version.

Maintaining a good signal to noise ratio is very hard without the transformer, but using either the NE5534A orthe dual version (NE5532 or [and IMO only if you really have no choice] LM833) is a good start. With an inputnoise figure of less than 5 nV / (see Noise Figure and Other Stuff) for an explanation), these are probably

http://sound.westhost.com/noisefigure.htm



the best choice for a reasonable price, and they are readily available - this is a bonus, as I hate to usedevices that are hard to get.

There are (allegedly) better opamps, but they may not easy to get or may be inappropriate for high qualityaudio, and will be more expensive as well. The NE5534 is a very good opamp, and is always a safe bet.Make sure that you use the NE5532A for the input amps, as these have a better noise figure than thestandard version.

Mic / Line Input Module (Electronically Balanced Input - Version 2)

This version is very similar to the one above (although it might not look like it), but uses an Analog DevicesSSM2017 microphone input amplifier. It is electrically almost identical to the "version 1" amplifier, but uses asingle device. It is a far cheaper option if you can get the SSM2017 devices, but I figured that mostconstructors will have trouble finding them (hence version 1).

Since this article was originally published, the SSM2017 has been declared "obsolete", and is no longeravailable. Texas Instruments make a pin compatible replacement, called the INA217, and its performance issaid to be as good or better than the original.

Figure 5 - Electronically Balanced Mic / Line Input Module (Alternate Circuit)

As you can see, this is a very simple circuit indeed. I cannot vouch for its performance, since I have beenunable to get the SSM2017, so the circuit is basically directly from the manufacturer's data sheet. The onlymodification is the input grounding, which is the same as shown for version 1.

The SSM2017 has been around for many years, so I must conclude that it is probably fairly good. Noiseperformance is something of an unknown, because of the rather obscure way it has been specified in thedata sheet. It is alleged that the noise figure is 950pV / (yes, that is pico-Volts), but this is only for a gainof 1000 (60dB). At lower gains, the noise figure climbs, and is 1.95nV / at a gain of 40dB and 11.83nV /

at a gain of 20dB.

Regardless, it can be assumed that the noise is very low, but I have also read reviews of products using itthat claim it is "ordinary". Exactly whether this is good or bad is unclear, but I would expect that you will notbe disappointed if you can get hold of the ICs.

Click on the PCB image to see Project 66 - yet another version of the mic/ line preamp - this one has a PCBavailable.

Tone Control Module




The tone controls are unusual in this design, because I wanted to have something a little more flexible thanthe standard 3-band EQ commonly used. As a result, there are two "gyrators" or simulated inductors, andone of these is made variable to allow the midrange control to be swept from 500Hz to 2000Hz. This isexpected to cover the range where most nuisance peaks and dips will be found, and will make accurateequalisation far easier than is the case with a fixed control. Bass and treble are conventional fixedfrequencies, but as can be seen are also connected unconventionally.

The channel insert jack allows a signal to be routed via any external device - compressor, graphic orparametric equaliser, or any of the multitude of effects that are now available. This is post EQ (after the tonecontrols), which is good in some respects (the external unit's noise is not increased by applying treble boost,for example), but is a problem if the channel insert is used to inject a signal into the mixer. This is not the wayany signal should be sent to the mixer, and I prefer it the way I have designed the circuit (but I suppose Iwould).

Figure 6 - Tone Control Module

The mid frequency control can be modified to change the range, so by multiplying the value of C3 by 4, thefrequencies are halved (so it will have a range of 250Hz to 1kHz) and dividing by 4 will double the frequency.Likewise, the bass and treble controls can also be modified to change the turnover points from 300Hz and2.7kHz respectively. With the values as shown (the mid frequency is set to 1kHz), the tone controlcharacteristics are shown in Figure 6a. As can be seen, there is plenty of variation, and bass and treblecontrols are deliberately moved away from the centre frequency band. Conventional tone controls tend to becentred around 1kHz, but the idea of providing bass or treble boost and cut from this frequency has alwaysseemed a trifle ridiculous to me.



Figure 6a - Tone Control Response Curves

The +/- 3dB frequencies for the bass control are about 300Hz, and 2.7kHz for the treble. The midrangecontrol is variable, and its Q ("Quality factor") increases as the maximum boost or cut is approached. At lowsettings, the Q is quite low, so the control is "self adjusting" to some degree. High Q values are normally notneeded at moderate levels of boost or cut, but if there is a real problem frequency (the rim of some snaredrums springs to mind), it can be notched out very effectively to get that "fat" sound without the hollow ring.

I have not included high or low pass filters in this design, but they can easily be added. Virtually any simpleopamp filter can be used, or even a passive first order (6dB / octave) could be included if you wanted to.

Fader And Auxiliary Sends

The fader is connected to the "tip" connection of the insert jack, so a signal can be directly inserted. Whennot in use, the switching jack connects the output from the tone controls directly to the fader. It is important touse the best quality faders that you can afford (or can find - they are pretty thin on the ground, I'm afraid).

The Aux 1 send can be switched pre or post-fade, while Aux 2 is post fade only. An additional switch can beused to allow pre and post fade for this as well, if desired. The PFL (Pre-Fade Listen) push button isdesigned to allow you to listen to the signal, even if the fader is fully off, and over-rides the main headphonesignal. This is done in the master module, and the PFL switch is connected to a bus.



Figure 7 - Fader, Channel Insert and Aux Sends

When more than one PFL switch is pressed, there is still a minimal drop in level, but at less than 0.1dB thiswill not be audible. To remove the problem altogether, use a buffer stage before the PFL button.

Optionally, the main output can be switched to one of a number of stereo buses, allowing the mixer to beused with group masters, all feeding the main stereo send. This is commonly used to allow sub-mastercontrol for a group of microphones, such as drums, back-up vocals, horn or string sections, etc.

Although this complicates the construction of the mixer (yes, even more), it allows an entire section to beraised or lowered in the mix with a single fader. Without this, if for example, the percussion section needed tobe a little louder for one song (or one section), it would be necessary to operate perhaps four or morechannel faders simultaneously. This can be done, but it is far easier if the sub-masters are used. A singlefader adjusts the entire section, maintaining the balance of the mix exactly as it was originally set up. SeeFigure 9 for the details.

The mic / line module can also be used as a recording mixer, by taking a send to the tape machine from thechannel insert or an extra connector could be used. If you use the channel insert, use stereo plugs with tipand ring shorted (so the mix is available in the studio monitors). For recording use, it would be a good idea tohave an additional (switchable) input for tape playback, wired in parallel with the XLR. If balanced outputs arerequired for the recorder, then you will have to wait for the master module, since this has a floating balancedoutput circuit.

Peak Detector

The peak detector is the best method of ensuring that all signals are well below clipping level. In this design,the detector operates when a signal is greater than 2V RMS, and is not polarity sensitive so it will indicate



when either a positive or negative peak exceeds the threshold. The dual opamp is a very basic (and cheap)1458 type, as it is more than adequate for this application.

Figure 8 - Peak Level Detector

The peak detector can be a simpler affair than that shown, but most of the simple ones only sense onepolarity. This is not suitable in my opinion, because audio signals can be extremely asymmetrical, so oneside can be clipping and you would not know it without listening carefully. Note that the circuit is designed toensure that no LED switching currents flow to the signal earth. These currents are "dirty", in that they containfast switching times and the resulting transients. It is important to ensure that the earth connection used isseparate from the signal earth, and must not run in parallel with the mixing buses or analogue earth (ground).

Note that the peak detector is after the tone controls. I have seen many circuits where the detector is beforethe EQ, and it is entirely possible to have a signal that just flashes the LED, but goes into clipping when EQ isapplied. This arrangement will hopefully help the user to avoid any such problem.

Although C3 is shown as 10uF, it can be as small as 100nF. Using a smaller cap allows shorter transients tobe captured, but also reduces the display time. I suggest that you experiment with the value to findsomething that suits you. My personal choice would be to use 1uF.

The detection threshold is set by the resistor string R1, R2 and R3. As shown, it is ±2.75V peak (closeenough to 2V RMS). Increase the threshold by increasing R2, and vice versa. The maximum I recommend isto increase R2 to 22k, giving a threshold voltage of ±5V - equivalent to +16dBm. The closer you get to theclipping threshold (+22dBm), the greater the risk of momentary overloads causing clipping.

Output Switching For Sub-Masters

The use of groups or sub-masters makes a live (i.e. stage) mix much easier. It is generally not needed forstudio work, but can still be very useful for the mixdown.



Figure 9 - Switching for Sub-Master Groups

A method of switching to multiple buses is shown in Figure 9, and uses a dual-gang rotary switch to selectthe required bus. As shown, you would be able to select sub-masters 1 to 4, the master mixing bus or Off,giving six positions in all. A 5 position switch would leave out the "Off" option - this is the version shown onthe panel artwork above. This (or an expanded version) provides the maximum possible flexibility for the finalmix.

Still To Come

In following articles, I will show the master module, power supplies and additional input modules. This is byfar the most complex single module, so it gets easier after this one.

Projects Index

Main Index

Copyright Notice. This article, including but not limited to all text and diagrams, is the intellectual property of Rod Elliott, and isCopyright (c) 1999/2002. Reproduction or re-publication by any means whatsoever, whether electronic, mechanical or electro-mechanical, is strictly prohibited under International Copyright laws. The author (Rod Elliott) grants the reader the right to use thisinformation for personal use only, and further allows that one (1) copy may be made for reference while constructing the project.Commercial use is prohibited without express written authorisation from Rod Elliott.

Updated 06 Feb 2000 - corrected errors in PFL switching and peak indicator send./ 13 Nov 99 - corrected error in tone control calculation




http://sound.westhost.com/project30b.htm[11.01.2010 19:15:04]

Elliott Sound Products Project 30b


Introduction

In this article, the Mixing Modules, Line Output stages and the start of the power supplies are described.There are two main types of mixing module - a stereo unit to accept the outputs of the Mic/Line modules, anda mono version for the Auxiliary sends that includes an Aux return as well (these are typically used foreffects, or as a simple foldback mix).

The number of modules depends entirely on the final configuration you are aiming for. A typical unit mighthave four stereo sub-groups, one master mixer and two auxiliary send mixers.

The master modules do not have tone controls, as it is anticipated that the final mix will be sent to anoutboard graphic equaliser or parametric equaliser for room acoustic tuning. Simple tone controls in themaster sections are worse than useless.

Description

A block diagram of a "typical" configuration is shown below. This is based on my original premise of foursub-groups, two auxiliary sends, and a master. All sub-groups outputs are re-mixed to the master mix bus,and optionally can be connected to balanced line output stages as well. The aux sends are not shown, butwork in much the same fashion (except they are mono, not stereo).

Figure 1 - "Typical" Configuration

As shown, channels 1 & 2 are switched to Mix Bus 1, channel 3 to bus 2, channels 4 & 5 to bus 4, andchannel "n" to the master bus. The outputs of each sub-master (or group) all connect to the master bus. Allbuses are stereo, but this is not shown for clarity. This is to demonstrate the flexibility of using the multiple



sub master groups, a 6 channel mixer with 4 groups and a master would not be worth the effort, but whenyou have a 36 channel unit this will change !

Mixing Modules

The mixer modules are "virtual earth" types, meaning that the actual mixing bus carries signal current, butthat little or no voltage is measurable. This is the most common type of mixer, as it means that there is nointeraction between the various input level controls (the faders).

The input capacitors should be high quality tantalum types, as these are capable of operating safely with nobias (or a slight reverse bias). The Gain control is used to trim the Master Module gain in the same way asthe input modules. This allows the operator to get all faders in a position where gain can be increased ordecreased by the desired amount, and returned to a known starting position. A setting of -6dB on all fadersis a good starting point, and this allows the operator to be able to change the level of any channel or group,and set the level back to the "standard" starting position again.

An output is provided for peak detectors (the circuit for these is in Part 1) - their use is recommended, but notmandatory. Even better is to use good quality meters (to be described in a later article) for all Sub-Groupsand Masters.

Figure 2 - Stereo Mixing Module (Groups & Master)

You will also need mixers for the auxiliary sends from each channel. Maybe you will decide to have morethan 2, but in any case you will require as many auxiliary mixers as you have sends. These are virtuallyidentical to the stereo version, but of course are mono. The circuit is shown in Figure 3, and is a dual mixingmodule.



Figure 3 - Mono (Auxiliary) Mixing Modules

Line Output Module

The line output stages (like the mic / line inputs) can be either transformer of electronically balanced. Bothtypes are described so you can choose the one you prefer (or can afford - good transformers are expensive). For recording work, the line output stages can also be connected to as many of the mic/line input modules asneeded to suit your recorder. Typically, line out modules would be connected to the output of the tonecontrol section for each channel. The fader is used only for mixdown, and to set the correct mixing levelduring recording. Some engineers prefer to record flat (no equalisation) to ensure that nothing is lost in theoriginal. Adding a switch to connect the line out pre- or post-equalisation allows for all possibilities.



Figure 4 - Active Line Output Module

The signal is reduced by 6dB as it is applied to the line out module, otherwise the level would be double thatexpected due to the balanced circuit. The transformer version (below) is unity gain, since transformers donot have this problem - they have others instead.

Figure 5 - Transformer Line Output Module

There are a great many different circuits for balanced line drivers, but the simple approach is often the best.The simplest is to use a transformer, but unfortunately, if you want good quality you will pay for it.

Click on the PCB image to look at Project 87 - Balanced Line Driver & Receiver - PCBs are available.

Power Supply

There are two sections to the power supply, the main +/-15 Volt supplies and the 48V phantom feed supply. The main supplies require considerable current capacity, due to the large amount of electronics involved, soconventional 3-terminal regulators will not work unassisted.

The 48V phantom supply needs to be as quiet as possible, as any noise will appear at microphone inputs. In




theory, the noise is simply cancelled because it is common mode, but I believe that a quiet power supply isworth the effort.

Figure 6 - Power Supply Transformers and Filters

The transformer and bridge rectifiers are shown in Figure 6 for the two supplies. The size of the transformerfor the phantom power supply depends on the number of phantom powered devices you want to operate. Since each will take a current of (typically) 7.0mA or 14mA worst case, we need to design for the worst. If 10phantom powered devices were to be used at once, we will need a supply capable of 140mA at 48V. Beforeregulation, we will need about 60V minimum, and I suggest a 20VA transformer which will be more thanenough.

For the +/- 15V supplies, I have used an estimate of 100mA per module (this is a bit of overkill, but it is farbetter to have it and not need it than to need it and not have it). Allowing for a 36 into 4 into 1 mixer with 4Aux sends and 20 line outputs, this means a total current of about 4A. To be on the safe side, a supplycapable of 5A is the design goal, so using a 20-0-20 transformer (40V centre tapped) at 1.4 times the DCcurrent means that we will need a 150VA transformer as a minimum. (In case you missed it, the AC currentinto a capacitor input filter is 1.4 times the DC current for a full wave bridge rectifier.)

The minimum power supply components are listed below, feel free to increase the ratings on transformersand capacitors, and adjust the fuse rating accordingly.

Transformer - 50V secondary, 20VATransformer - 40V centre tapped, 150VAElectrolytic Capacitors, 4 x 4700uF 50VBridge Rectifier - 5A 200VBridge Rectifier - 25A 200VMains Fuse - 1A slow blow

I stronly recommend that the power supply be built in a separate box, so that no transformer hum is injectedinto the circuitry. The virtual earth mixing buses are particularly succeptible to magnetic flux leakage fromtransformers, and a separate supply will eliminate this problem, and helps to keep the weight down as well. The power supplies will require generous heatsinks, so a small enclosure is not an option.

Still To Come

Power Supply regulatorsMonitoring amplifiers



Pre Fade Listen circuitryMeter bridgeGeneral layout of the complete unit.Phono and auxiliary input modules

Projects Index

Main Index


Created and Copyright (c) 23 Oct 1999 Rod Elliott




http://sound.westhost.com/project30c.htm[11.01.2010 19:17:16]

Elliott Sound Products Project 30c


Introduction

Stage 3 shows the power supplies for the mixer. The supply needs to be quite substantial, due to the highcurrent drain of a complete 36-4-2 mixer. I will have to leave it up to individual constructors to determine ifthey really need this much power, or can survive on a lesser version, but if the full power supply is built, it willallow for expansion later. The phantom supply is designed to provide 150mA at 48V, which should be morethan enough for any application.

Also included is the headphone power amp (two needed for stereo), with the mixing and selection coming inthe next section.

15 Volt Power Supplies

The mixer requires +/- 15V at up to 4A for a fully configured system. Since this is well outside the capabilityof a 3 terminal regulator, I have used booster transistors (a quite common thing to do) to increase theavailable current. The regulation is completely controlled by the 3-terminal regulator, and the transistorsimply boosts the current. Dissipation in the transistor will be quite high, so good heatsinking is essential.

Using a 20V (AC) transformer gives about 28V before regulation, allowing plenty of margin for low supplyconditions, but increases the transistor dissipation. At 4 Amps output, worst case transistor power will beover 40W, so the heatsink will need to be rated at less than 1°C /W to ensure that the transistors remain wellwithin their safe operating area.

Figure 1 - Transformers and Rectifiers

The transformers, rectifiers and capacitor banks in Figure 6 are also shown in the previous section, and the



circuit is included here for completeness. As indicated in the last section, T1 needs to be at least 150VA, andT2 can be a smaller (and cheaper) 20VA. The bridge rectifiers need to be more substantial than you mightthink, because of the sustained current. I suggest that a 25A bridge is used for the +/- 15V supplies, and a5A bridge for the 48V supply.

Figure 2 - 15V Regulators

Figure 2 shows the regulators for the +/- 15V supplies. DC from the rectifier is supplied to the 3 terminalregulator IC via a 33 Ohm resistor. When the current exceeds about 20mA, the power transistor will turn on,and the IC will ensure that the DC output is kept exactly to the specified voltage. This ensures that theregulator IC is operating at a low power (requiring only a small PCB mount heatsink), and will remain cool.

Typically at an output current of 4A, the transistors will require about 200mA of base current, which must bepassed by the regulator. The maximum regulator power is therefore about 2.6W - a 10°C/W heatsink istherefore sufficient (but only just - use a bigger one if possible). Make sure that each regulator is on its ownheatsink, and uses no insulating washer for maximum heat transfer. Use heatsink compound betweenregulator and heatsink. Do not mount the regulators on the main heatsinks - these will operate at a highertemperature than the small individual heatsinks.

The power transistors will be operating at a sustained high power level, and must have substantialheatsinks. Dissipation will be in the order of 25W each at 4A, which means that the temperature must bekept below 100 degrees. Using the arrangement shown with parallel transistors, this is easily achieved, andwill require a heatsink with 1.0 degree C/W for each transistor pair. By this means, a smaller heatsink maybe able to be used than would otherwise be the case, since the dissipation of each transistor is reduced andthe junction to heatsink thermal resistance is also reduced. The heatsink will still run at quite a hightemperature, and should be mounted where it cannot be touched, but still has good airflow. I recommendeither the biggest heatsink you can accommodate, and/or use a fan to assist cooling. The transistors shown



are the minimum specification you can use - preferably, use higher power transistors.

The diodes (1N4001 or equivalent) around the circuit ensure that disconnection of the DC input will notdamage the regulator. Note that the +ve and -ve regulators have different pinouts - do not get them mixedup, or they will be damaged or destroyed when power is applied. Please check the datasheet to make surethat you get the IC connections right!

Note also that there is no easy way to provide short circuit protection to this configuration, and care isneeded to make certain that a short cannot occur. For this reason, I suggest that the transformers, rectifiersand capacitors are housed in a separate case (typically floor mounting), and that the DC output be fused asshown in Figure 1.

The regulators can be housed in the main mixer, keeping all input wiring well away from mix buses. The DCinput will have ripple at high current, and if allowed near a bus will inject hum into the system. The 100uFcapacitors should keep the circuit stable at all operating levels, and can be bypassed with 100nF caps ifdesired.

48 Volt Phantom Supply

Although high voltage 3 terminal regulators are available, they are not readily available to most constructors. I have therefore elected to use a discrete design, which although quite simple will give very good results. The regulation does not need to be any better than about 1 to 2% from no-load to full-load, but noise (ripple)must be kept to a minimum.

Figure 3 - 48V Phantom Supply

At full load, the power in the series pass transistor (Q2) will be only about 1.8W, so a massive heatsink willnot be necessary. A small 10 degree/W sink will ensure that the maximum temperature rise is less than 20degrees Celcius, which is a generous safety margin. Q2 should also have a heatsink, as its dissipation willbe about 400mW at no load. A simple flag heatsink will suffice.

Using a resistor filter is awful for regulation, but gives good ripple rejection, and is simple and cheap. A 10Ohm resistor and 2200uF capacitor will have a very profound effect on ripple applied to the regulator, whichsimplifies its design. This simple addition reduces ripple to about 1/30th of that without the filter, so is wellworth the small extra outlay for the resistor and cap. The circuit shown has better than 80dB of ripplerejection, so the output will typically have less than 20uV of hum. As this is applied as common mode to themicrophone leads, the overall rejection will be found to be more than adequate. The circuit in Figure 3 has aregulation of better than 2% from no-load to full-load, which is quite acceptable.



When finished, adjust the trimpot (R6) to give an output of 48.5V with no load. This will fall under full loadconditions (140mA) to about 47.8V or so, indicating a regulation of about 1.4%.

Click on the PCB image to see Project 96 - a (slightly) different version of the phantom feed power supply. PCBsare available.

Headphone Amplifiers

The headphone amps are based on the LM1875 power opamp. This is capable of 25W, but in thisapplication is deliberately restricted to 0.5W into 8 Ohm 'phones. This is more than enough, but at leastensures that the amp will never clip at any listening level that is even remotely sensible.

Figure 4 - Headphone Power Amp

The units can be assembled on Veroboard (but see below first - there is a PCB available for this power amp).The 100nF caps in the supply lines need to be as close to the power pins as possible. Some care is neededwith the layout to keep input circuits well away from the output, as these devices have wide bandwidth andwill oscillate if construction is unsuitable.

The 56 Ohm output resistor should be at least 1W, and limits the power to the headphones. Two circuits willbe needed for stereo 'phones, and the ICs should be mounted on a suitable heatsink. Make sure that thepower rails for these amps are taken straight back to the regulator outputs, and not connected to the mainsupply buses for the mixer modules. Relatively high peak currents with unpleasant waveforms may otherwiseintroduce distortion into the completed unit.

The Pre-Fade Listen and headphone selection and mixing circuitry will be published in the next installment.

Click on the PCB image to see Project 72 - almost identical to the above, and PCBs are available.

Still To Come

Headphone selection and pre-fade listen amps. Also the talkback mic input, phono and auxiliary inputmodules, metering circuits, and a few miscellaneous other bits.





There will also be an overall layout diagram, and a picture of what the panel of a complete system might looklike. This in itself should be interesting, as I have no idea yet how I am going to make such a large drawing(and keep the file size down to something sensible).

Projects Index

Main Index


Update Info: Page created 20 Nov 1999./ Updated 02 Feb 03 - corrected LM7x15 regulator pinouts




http://sound.westhost.com/project30d.htm[11.01.2010 19:19:10]

Elliott Sound Products Project 30d


Stage 4 - PFL, Monitoring, Talkback Mic & Auxiliary Modules

Introduction

This installment describes the pre-fade listen and other headphone mixing and switching, as well as thetalkback mic amp, phono and auxiliary input modules.

The pre fade listen (PFL) is designed to override the main signal, so although an individual channel is heard,it is not in complete isolation. This is useful, because the recording engineer hears some of the main signalso not only is the channel heard, but it is in context. This is adjustable.

The talkback microphone is comparatively simple, as it does not require phantom feed, phase reversal orattenuator pads. It still has adjustable gain, and uses the same mic preamp as used in the channel modules.

The phono and auxiliary modules are similar to the main channels, but are stereo, and have no mic preamp.The phono preamp is based on one of my other projects (surprise), and can be added to as many channelsas desired - or not used at all.

Monitor Switching

The headphone monitor switching is fairly simple, and basically selects any of the output buses as thesource. If the bus is mono (such as auxiliary sends), the mono signal is sent to both channels of theheadphone amp.

The switch is used to select the Aux Sends (1 or 2 - more if installed), Sub-Masters 1 to 4 or the Master bus.While the selector is used for long-term monitoring of a source, it can only be over-ridden using a PFL (alsocalled 'solo') button anywhere on the desk when switched to PFL. The selector switching is shown below -like all switching, it is very simple. The inputs come from the mixing amplifier for each source indicatedexcept for the PFL (Pre Fade Listen) which comes directly off the PFL bus.



Figure 1 - Monitor Switching

PFL and bus monitoring is completely conventional, except that a small amount of the final mix can beinjected into the PFL bus to allow each channel to be heard in context rather than complete isolation. Thismay be preset to any level desired (including off).

If my original configuration is used (as shown above), you will need a dual gang 8 position rotary switch forselection. If more aux sends (or sub masters) are used, the switch becomes bigger - the largest switch youwill be able to get easily is a 12 position.

An alternative is to use interlocked push-button switches, but these are fairly expensive and may take up toomuch space on the panel.

Pre Fade Listen

The PFL bus normally (almost) floating - i.e. not connected to anything. It is only connected to the inputchannel(s) where the PFL button is pressed. One variation is that it can have a selectable amount of themain (Master) mix present at all times, but only in mono. When a PFL button is pressed, this signal isreduced so the channel signal is heard in context, and the combined signal is applied to the headphones.

It is possible to use more than one PFL button at a time, so that two or more channels can be monitored atonce.



Figure 2 - Pre Fade Listen Bus

The PFL switches are shown as a reference only. As can be seen from each module, the PFL buttonconnects the channel to the bus via a 1k resistor. Since the feed from the Master outputs (via VR1) uses 22kresistors, the PFL signal will be 10 times that of the Master 'background' signal with the level pot set tomaximum. If you don't want to use this, then the PFL bus is simply connected to the monitor switch as shownabove.

Talkback Microphone

The talkback mic can be switched to any of the mix buses, and also has its own separate output. This outputis always active (as long as the talkback switch is activated), regardless of the bus select switch setting. Theoutput may be balanced, using the selected balanced output driver from Stage 2. There are no tone controlsfor the talkback mic, but these can be added if you want.

The connection to the mix buses is a fixed 50/50 split - there is no pan control. The signal is sent to bothchannels of the selected bus whenever it is activated.

Figure 3 - Talkback Microphone



The high pass filter may be omitted if you don't want to use it, and it can be fixed at 80Hz by removing thedual-gang pot. The series resistors must remain, but are changed to 56k, and the pot is simply shorted out. Ifyou want, you can use the filter with any (or all) of the mic/line channels. At maximum pot resistance, the -3dB frequency is 15Hz - increase C1 and C2 to lower this (double the capacitance is ½ the frequency.

The full range using a 100k pot as shown gives a minimum of 15Hz and a maximum of 160Hz.

Auxiliary Input Module

The auxiliary input module is line level only, and is unbalanced. This is intended for inputs from CD players,phono cartridges, tape machines and effects returns. Each is stereo, and you can use (or not) the phonocircuit as needed. The tone control circuit is a stereo version of the main mic / line channel controls - it is notshown below. The tone controls (if desired) are inserted at the points marked 'TONE'. The fader is a simplestereo affair, and there is a balance control rather than a pan pot.

The line input stage is a simple variable gain amplifier, having a gain range from -10dB up to a maximum of20dB. This will be more than sufficient for typical sources.

Figure 3 - Auxiliary Module

The outputs are connected to the mix buses using the switching shown in Stage 1, Figure 9. This may besimplified (or direct connected to the master bus) if multiple bus selection is not required.

Phono Preamp

The phono preamp may be any of those on the ESP website, but for the best performance overall, I suggestProject 06. This is a very high quality phono preamp. It is unlikely that more than one phono stage will beneeded (if that), but for the cost (which is minimal) it is worth including.

The phono preamp should be switch selected on an Auxiliary module (as shown in Figure 4), and connected




to the input of the amplifier shown. In this way, the module is not dedicated to a single task. It is also possibleto have the phono stage as a separate unit (but inside the mixer chassis), and connected to the desired auxchannel using a patch lead.

Click on the PCB image to see Project 06 - the recommended phono preamp to use with the line input module -PCBs are available.

Projects Index

Main Index


Update Info: Page created 17 Jan 2000




High Quality Sound Mixer

Documents

Transcript of High Quality Sound Mixer