Matrix Mixer

Introduction
A topic that's been raised a few times now is a matrix mixer. So, you may well ask, what's a matrix mixer when it's at home? The ideal is that you have a number of inputs (typically from 3 to 16), and a number of outputs - there may be more (uncommon) or fewer (much more common) outputs as there are inputs. An array of pots allows you to mix the output of any input channel into any output channel, allowing anything from (say) 6 in to 1 out to 6 in to 6 out. Matrix mixers are also known as crosspoint attenuators in some cases.
As an example, output 1 might be set to have only the signal from input 3, while the outer outputs have a mixture of any two or more inputs. Naturally, it can be set so that each output contains only the signal from its corresponding input, but this is not the way such a mixer is generally used.

Example Of A 6 x 6 Matrix Mixer The above picture is an artist's impression of what may be a typical matrix mixer. All inputs and outputs would normally be at the back, but if it's more convenient for your use, they can just as easily be at the front or the sides. The white areas shown are so you can write a description for each output - it is also useful to have the same for the inputs as shown. You can use small pieces of magnetic whiteboard material if you use a steel chassis.
While it would seem that this would require a lot of circuitry, in reality the circuit is remarkably straightforward. There is more repetition than actual complexity. Because inputs and outputs are arranged in a matrix, it's only necessary to provide a level control (a pot, usually rotary) that allows any input (or number of inputs) to be mixed into any output (or number of outputs). The general scheme is shown in Figure 1, and as you can see there really isn't a lot involved. There are a lot of pots though, depending on your requirements.
 As you can see, there are lots of pots, and it is very likely that the pots and knobs will be the most expensive part of the project. Although each input and output is shown directly connected, this isn't the way it's done in practice. All inputs must be buffered with an opamp that has enough output capability to drive all the pots for that row (or column, depending on how you arrange it), and the outputs need to be fed into a proper mixing stage. These will be described in more detail below. All pots should be 100k linear (not log!), and mixing resistors are 15k as shown.
One common use for matrix mixers is to provide foldback for the players on stage, but there are many other instances where you might wish to mix a number of source signals into several different mixes. Matrix mixers are also used in some synthesiser designs, either for control voltages or audio signals. The unit featured here is strictly audio. The uses for a matrix mixer are many and varied, and if you need one, you only have two choices - buy one that provides the functionality the manufacturer thinks you need, or build one with the functionality you actually need.
Inputs and outputs can be used for anything you like. Add headphone amps and you have as many headphone outputs as you need, each with its own mix. You can include a switch (or individual push-buttons) to allow you to monitor each output independently - the possibilities are endless. The matrix does not need to be "square" as shown (same number of inputs and outputs), and you can have as many inputs and as few outputs (or vice versa) as you need.
The ultimate limitation is the load presented to the input buffer amplifiers, and the noise gain of the mixers. These topics are covered in detail below.
Another use for matrix mixers is for mixing control voltages (for example 0-10V lighting controls or synthesiser control voltages), but as shown this unit is strictly for audio. While it would be possible to make it completely DC coupled, this is of little use for the vast majority of applications. DC operation causes problems if even small DC voltages are present across any pot(s), as it makes them sound "scratchy", so the present unit is designed for audio only.

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Description The driver/buffer and mixer stages are based on the P94 Universal Preamp/ Mixer board (see Project 94 for specific details). Essentially, each input is an inverting buffer, and as shown each has a gain of 2 (6dB). This increases the level, so helps to keep the signal to noise ratio better than it might be otherwise. It is important that the signal remains well below clipping level at all times, so higher gain is not recommended unless you are certain that the applied signal can never exceed that voltage which will cause the buffers to clip.
The other consideration is the load on the opamp. Many popular opamps can't drive a load less than around 2k before the maximum output level is reduced and distortion increased. Opamps such as the NE5532 or OPA2134 can both drive 600 ohms at more than ±10V peak, and either of these is almost essential for the input buffer stages. If we allow a sensible safety margin and limit the maximum load to ~1k, either opamp can drive about 12 sets of pots (12 outputs), allowing for the worst case where all pots are set to maximum level.
The general arrangement for an input buffer is shown below. You need as many of these as you have inputs, so a six input matrix requires six input/ driver stages. This is the first section of the P94 board, with the second section being optional. It's not strictly necessary, but without the inverter stage the whole mixer will be inverting. Contrary to popular belief, this generally doesn't matter, but it can cause problems if the matrix mixer is used in conjunction with a normal mixing desk. In general, it's better to ensure that mixers are non-inverting.

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Figure 1 - General Electrical Layout Of A Matrix Mixer-Figure 2 - Input Gain Stage & Inverter-Figure 3 - Mixer & Output Stages

 

 

If you don't care about inversion, the second stage can be omitted. As noted above, I don't recommend this, but it is a choice you can make if you know that it won't cause problems elsewhere. One such problem may be having the guitar signal disappear almost completely because you have mixed equal levels of inverted and non-inverted guitar signal ... and yes, this can happen, although it's fairly unlikely. If you wanted to, you could also include the tone controls for each input channel, but I expect that this would be unwanted in the vast majority of cases.
The schematic above shows the component designators for the P94 board. Parts showed greyed out are not used. The connections for the treble control (T1, T2 and T3) are bridged together, and C104/205 (second channel) are replaced with wire links. All other component locations must be left vacant.
    Parts shown greyed out are not used, and R111 is replaced with a wire link, or a 10 ohm resistor. The latter may be helpful if the unit is likely to be used where there is considerable high frequency electrical noise. The on-board mixing resistors aren't used because there aren't enough of them for most applications, and it's better to physically wire the pots more or less as shown in Figure 1, using busses wherever possible. You need one mixer stage for each output channel, and I suggest that you limit the number of inputs to 12 or less, or noise will become a major problem. Parts marked with a * are not provided for on the PCB, but are separate. The 100pF capacitor may be soldered directly to the leads of R115/R215, either on top of or underneath the PCB. This cap reduces any high frequency problems that may be created by the mix bus capacitance.
The same opamps are also recommended for the mixers, because they are very quiet. There are quieter opamps, but the NE5532 is ideally suited here, because the effective input impedance may be quite low. Low impedances suit bipolar transistor input opamps best, so be wary of "low noise" FET input opamps. With all pots for an output set to maximum (or minimum), the input resistance may be as low as 2.5k, so the opamp will have a worst case noise gain of 6 with the configuration and values shown.
For those who don't know the term, noise gain is the gain the stage has for its own internal noise, plus any thermal noise from external resistors. In the case of a mixer stage, it's very easy to have a noise gain of 10 or more, but a signal gain of less than unity. This degrades S/N ratio rather badly, because the opamp has much higher gain for the noise voltage than it has for the signal. For this reason, it's normal to operate any mixer at the highest practicable level, and a matrix mixer is no exception.
Note that with the output level control shown, output impedance is comparatively high, and is not suitable for driving long signal cables. If this is a requirement, then you will need to add buffer stages to all outputs, after the final master volume control. These could be the ESP Dual Balanced Transmitter circuits (Project 84) if balanced outputs are desired. In some cases, the outputs will be used solely to drive headphones, in which case the ESP headphone amp (Project 113) is a good choice. Otherwise, The first stage only (U1 and associated components) of the p94 board can be used for the output buffers as well. If you do use this arrangement, leave out R117/217 (after the master volume pot) or it will be duplicated by the input resistor on the P94 stage.
You will also need a power supply, which may need to be quite robust depending on the size of the mixer. Excluding any balanced outputs or line drivers, a 6 x 6 matrix as shown here will draw up to 90mA at ±15V, assuming 5mA for each opamp (10mA for each package). Each input section uses one opamp (half of each of the 2 opamps), and each basic output stage uses half an opamp. This can be supplied using the Project 05 power supply, with generous heatsinks on the regulators.

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Construction Construction without the PCBs is not really recommended because of the number of stages needed. Without the boards, there is a very high risk of problems caused by wiring mistakes, but the boards make the unit somewhat modular, and faultfinding is much easier should something go awry during construction.
The next drawing shows the way that the busses can be connected. This is an example only, so if you have a preferred way to do it, then by all means do so. Note that the earth wiring (GND, shown in green) should be connected (soldered) to each pot body if at all possible. This ensures that the busses are held firmly, but just as importantly, provides a solid earth connection to each pot in the matrix. Mix busses ideally need to be kept away from the chassis to minimise stray capacitance. Even a small amount of stray capacitance can cause the mixer opamps to have significant treble boost at high frequencies, possibly leading to instability or greatly increased noise and distortion. The extra cap added to the mixer will prevent most problems, but good wiring practice is essential regardless.
If the mix busses require additional support, this can be provided by using cable ties and suitable pieces of insulation as required. For example, if a short length of heatshrink tubing is placed over each 15k resistor, these may then be cable-tied to either the earth bus or the input bus. Remember that all busses need to be strong enough to survive the kind of usage that the mixer will suffer in normal life. There is no place for flimsy construction that may be alright in the studio, but fails on its first field trip.

Figure 4 - How To Wire Busses To Pots This unit must be housed in a metal enclosure, because the mixing busses in particular are rather sensitive to any noise pickup. Because the unit is a matrix, most of the wiring can be done using bare tinned copper wire, with pot connections bent up or down to ensure that the busses can't short out to each other. If you use this method, make sure that the wire is thick enough to ensure that it can support itself - even during rough handling. If done properly, this bus structure is very stable, reliable and was common in the early days of mixers. Most matrix mixers will be comparatively small, so the relatively short runs of self-supporting tinned copper wire are easy to work with. Even for a 12 x 12 matrix (and that's a mighty big matrix mixer), the busses will be no more than ~350mm long, and are supported at every pot they pass.
One trick that's worth knowing ... how to make a length of tinned copper wire perfectly straight, with no bends or wrinkles. Cut the length you need for a bus, plus about 50mm extra. Hold one end in a vise, and the other with a pair of pliers, and give the wire a short, sharp pull. This stretches the wire slightly, and leaves it nice and straight so it looks neat when placed into position. The ends that were held will be damaged, so these bits are cut off before you install the bus.

Surround Sound

Introduction
This surround-sound decoder is based on the "Hafler" principle, first discovered by David Hafler sometime in the early 1970s. The original idea was to connect a pair of speakers as shown in Figure 1, for use as the rear speakers in the surround setup.
This is ok just as it stands, but problems are created if the main speakers are bi-amped or using bridging, for example, since there is no longer a full-range / full power signal available for the rear speakers. There is also no way to control the level reproduced, since it will always simply be the difference signal between left and right channels.
If the signal is mono, then the signal in both channels will always be more or less identical, and there will be no output from the rear speakers at all.

Figure 1 - The Original "Hafler" Surround-Sound Matrix

This circuit works by allowing the rear speakers to reproduce only the difference signal between the left and right outputs. All stereo encoded material has some difference between left and right channels (if it didn't, it would be mono), and it is this difference signal that is reproduced by the rear speakers.
It is important to ensure that the connection between the rear speaker negative terminals is not earthed, or they will simply be in parallel with the main speakers.

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Line Level Passive Version So, if you want to use separate amps for the rear speakers, basically you can't - unless you get sneaky. The first circuit in Figure 2 is completely passive, but requires that a suitable transformer is available. A suitable transformer means a line level, 10k impedance unit with a 1:1 ratio - these are scarce, but are available after a search.
You might be able to get away with a 600 Ohm unit, but because of the impedances you need, its performance will be very ordinary, with an extreme lack of bass (there is not enough inductance for a 600 Ohm transformer to work satisfactorily at high impedances). Loading the transformer will give back some of the bass, but the preamp is unlikely to be very happy with the resulting impedance. Having said this, I have used telecommunications transformers for this application (600:600 ohms) and they seem to work fine.

Figure 2 - A Passive Line Level Hafler Matrix Decoder

The circuit shown is not a bad compromise, although the impedances are too low for anything other than a solid state preamp (preferably using opamps). Using a telephony transformer (600 Ohm), the loss overall is about 3dB, with a low frequency -3dB point around 100Hz. This will vary depending on the quality of the transformer used, so experimentation will be needed. Although 600 Ohm telephony transformers are reasonably readily available, some of them are pretty ordinary.
My tests were on a really good one, built by an Australian company called Transcap. I think I can say with some certainty they will be rather unwilling to sell one-off quantities. Another manufacturer of really nice transformers is Midcom in the US, but you will have the same problem with them. These manufacturers are set up to deal with large orders from other companies, not the likes of you and me wanting one ("You want ... ONE ??") transformer. As a result you will have to take whatever you can get.
Since it is unlikely that this will be viable for most constructors, the alternative is to go active, using a dual opamp to perform the functions. This is described next.

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The New Circuit The schematic shown in Figure 2 is a simple way to achieve the same thing (with some additional benefits) at line level (i.e. before the signal reaches the power amplifiers - in a biamped system, this circuit must be between the preamp and the electronic crossover). The extras available are readily apparent ...

• Wiring is simplified (although additional power amplifiers are needed)
• We now have a centre channel signal available
• Provision for a mono signal to a sub-woofer is easy

Figure 3 - The Schematic of an Enhanced Hafler Matrix Decoder

Although there have been similar circuits published over the years, this is a little different in a few areas. I wanted to avoid having any active electronics in the main Left and Right channels, since this eliminates any possibility of sound degradation due to the introduction of the opamps. The input impedance of 50k will not pose a problem for any preamp (including valve types), and the main signal is simply in parallel with the additional circuitry.
No volume control has been included, since you already have one in the preamp, and the power amp used for the rear channels will almost certainly have a level control so the front/ rear levels can be balanced. Another volume control would just become another component to fiddle with, and since it would be rarely used, would probably become noisy over time, just from sitting in the one position permanently.

Note: When using the circuit shown above, the rear speakers should be wired out-of-phase. One speaker connects to the amp normally (red and black terminals used in the usual way), but the other should be connected with the speaker leads reversed. The difference is usually subtle (given that the surround information is often of dubious quality), but to get the full effect it's better to use the out-of-phase connection or you will not get the maximum benefit. This connection creates the R-L and L-R signals.

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How It Works
Opamp U1A is connected as a subtracting amplifier. Should the same signal be applied to both inputs, the output is zero. As a result, it will remove all common information from the stereo signal, and reproduce only the difference signal - in exactly the same way as the original Hafler design. U1B is a simple summing amplifier, and the output contains all the information from both the left and right channels. A possibility that springs to mind is that we could then subtract the difference information from this output, so that only material that is absolutely common to both channels would be reproduced. Would this improve the performance to the extent that the extra circuitry is warranted? I tend to doubt it, but may look into this further.
Centre Channel Control
The pot (VR1) is to set the centre channel level. This can be a trimpot, or a conventional pot mounted at the rear (to help prevent "fiddlers" from mucking up the settings you like). I have seen circuits which do not include this, which seems basically a bad idea. When the two channels are summed, the centre channel will typically have a level of -3dB relative to the left and right channels - provided the signal is not mono. Centre channel speech (for example) will be mono, so the level will be equal to that of each of the main speakers. Since the centre channel amp and speakers are rarely as powerful as the main Left and Right channels, there is a distinct possibility of overload of the amp, the speaker or both.
Since the centre channel is supposed only to fill the "hole" and provide a stable centre sound image, it does not need to be as loud - especially since it will almost certainly have inferior sound quality to the main speakers and will therefore degrade the overall sound quality. The level control will allow you to set the level to just sufficient to provide a stable sound image, and no more. In my system, I do not use a centre channel, and doing so would have an overall adverse effect on sound quality. If you have good main speakers and have a stable and well defined soundstage, a centre speaker is likely to do more harm than good.
The capacitor (C1) is optional. It provides a nominal 8kHz roll-off frequency (which is apparently quite normal for "real" surround-sound processors). This helps to minimise any disturbance to the main stereo signal, but feel free to leave it out, since most centre channel speakers probably won't be able to reproduce much above this frequency anyway.
Sub-Woofer Output
The sub-woofer output is simply taken directly from the centre channel mixer, and I included no low-pass filter because I don't know of any sub which does not have a filter already. Adding another one simply adds unnecessary complexity, and will introduce phase shift at the output that a phase compensation circuit (often included in sub woofers) may not be able to cope with.
Miscellaneous
The 100 Ohm resistors in the outputs are to prevent the capacitance of the signal leads causing the opamps to oscillate. At this value, they will cause no high frequency loss, unless you insist on 100m long signal leads (in my experience, these are uncommon).
It will also be noticed that there are two outputs for the rear speakers, simply in parallel. I included this because it is easier to wire if the user is connecting a stereo amp for the rear speakers. Naturally, a mono amp will do just fine, as long as it is capable of driving the two rear speakers in parallel (but out of phase with each other). This may not be possible if the speakers are 4 Ohm types (these are becoming more common in hi-fi, so its not that silly). If you do have 4 ohm speakers, you can connect them in series. To obtain the out-of-phase connection, join the red terminals, and connect the two black speaker terminals to the amplifier's output.

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Construction The unit can be housed in any suitable metal case (it does not need to be displayed, and can hide up the back of the cabinet). A metal case is preferred to prevent any noise (especially hum) pickup from mains cables, etc. To power the unit, I suggest the power supply presented in Project 05 - this is simple, safe and cheap.
Since heat generation is not an issue, the case can be as small as you like, as long as there is enough room for the RCA connectors and other components. Be careful that you don't pack everything in the case too tightly, though, because you might create short circuits jamming everything in if it is too small.
The dual opamp and other components can be wired on a piece of Veroboard (or similar), and layout is not critical. As always, use 1% metal film resistors throughout, for minimum noise and maximum stability and reliability. It is expected that this circuit will be extremely reliable as long as care is taken when building it, so there is no need to make it so it is easily serviced. I have built similar devices in the past that have never required repair in over 20 years.
All RCA connectors can be hard wired, and since crosstalk is not likely to be a problem with this unit, the wiring is not critical. You must pay close attention to earthing - all RCA sockets and the power supply centre-tap (0Volt line) must be connected securely to the case to prevent noise pickup.
If desired, 100nF polyester caps can be connected in parallel with the 100uF supply bypass capacitors, but they are not really needed and performance will not suffer if they are omitted.

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Delay Line One thing that is missing from this simple circuit is a delay line. This is normally used to delay the sound supplied to the rear speakers, and effectively makes the sound stage larger by making the rear speakers sound further away. It also adds a degree of additional ambience, and is used in virtually all commercial surround decoders.
Unfortunately, although a delay line project was published some years ago, the IC disappeared within minutes of publication - it is no longer made or available. But ... all is not lost .
A new one (which includes the delay IC and the PCB if you purchase the project board) is available - see Project 26A for all the details.

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Skema Surround Sound

HOME THEATER 2.1

HOME THEATER  2.1



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HOME THEATER  2.1
PCB
SURROUND SOUND DECODER
FRONT CH  [FR FL]
REAR CH [RR RL]
CENTER CH [C] 
SUBWOOFER CH [SW]

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Audio Amplifier Output Relay Delay

This is a simple circuit which I built to one of my audio amplifier projects to control the speaker output relay. The purpose of this circuit is to control the relay which turns on the speaker output relay in the audio amplifier. The idea of the circuit is wait around 5 seconds ofter the power up until the speakers are switched to the amplifier output to avoid annoying "thump" sound from the speakers. Another feature of this circuit is that is disconnects the speaker immediately when the power in the amplifier is cut off, so avoiding sometimes nasty sounds when you turn the equipments off.

Circuit diagram:

Audio Amplifier Output Relay Delay Circuit Diagram

Component list
C1 = 100 uF 40V electrolytic
C2 = 100 uF 40V electrolytic
D1 = 1N4007
D2 = 1N4148
Q1 = BC547
R1 = 33 kohm 0.25W
R2 = 2.2 kohm 0.25W
RELAY 24V DC relay, coil resistance >300 ohm

Circuit operation:

Then power is applied to the power input of the circuit, the positive phase of AC voltage charges C1. Then C2 starts to charge slowly through R1. When the voltage in C2 rises, the emitter output voltage of Q1 rises together with voltage on C2. When the output voltage of Q2 is high enough (typically around 16..20V) the relay goes to on state and the relay witches connect the speakers to the amplifier output. It takes typically around 5 seconds after power up until the relay starts to conduct (at absolute time depends on the size of C2, relay voltage and circuit input voltage). When the power is switched off, C1 will loose it's energy quite quickly. Also C2 will be charged quite quickly through R2. In less than 0.5 seconds the speakers are disconnected from the amplifier output.

Notes on the circuit:
This circuit is not the most accurate and elegant design, but it has worked nicely in my small home-built PA amplifier. This circuit can be also used in many other applications where a turn on delay of few seconds is needed. The delay time can be increased by using bigger C2 and decreased by using a smaller C2 value. Note that the delay is not very accurate because of simplicity of this circuit and large tolerance of typical electrolytic capacitors (can be -20%..+50% in some capacitors).

Transformerless Power Supply

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Transformerless Power Supply

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Bass Booster

Bass Boost is today's sound... whether it's the driving, gut-vibration pulsation of disco, or the solid bass line of soft, hard, or laid-back rock. One way to get the modern bass-boost sound without running out and buying an all-new expensive piece of equipment is to use a Bass Booster between your guitar, electronic organ or what-have-you, and the instrument amplifier. A bass booster strips the highs from the instrument's output signal and amplifies low frequencies, feeding on "all-bass" sound to the instrument amplifier. Naturally, the bigger the speaker used with the amp, the more powerful the bass: use 15-inchers with the Bass Booster and you can rattle the windows. Bass Booster is powered by an ordinary 9 volt transistor radio battery. It can be assembled on a small printed board or on a veroboard using point to point wiring. The booster connects between your instrument and its amplifier through two standard RCA Jacks.

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Bass Booster

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Parts:

P1 = 50K
P2 = 100K
R1 = 22K
R2 = 470K
R3 = 47K
R4 = 10K
R5 = 470R
R6 = 1K
Q1 = 2N2222
C1 = 2.2uF-25v
C2 = 100nF-63v
C31 = 00nF-63V
C4 = 3.3uF-25v
C5 = 470uF-25v
D1 = 5mm. Red Led
Q1 = 2N2222
B1 = 9v Battery
J1 = RCA Audio Input Socket
J2 = RCA Audio Output Socket
S1 = On-Off Switch

Using Bass Booster:

Connect your electronic guitar or other electronic instrument to input jack J1; Connect output jack J2 to your instruments amplifier's normally-used input. With power switch S1 off, key S2 so the instrument feeds directly to the instrument amplifier. With P2 set full counter-clockwise (Off), turn power switch S1 on, key S2 once, and advance P2 for the desired Bass Boost level. To cut back to natural sound just stomp down on S2 and key the Bass Booster out. Don't worry about leaving power switch S1 on for several hours of a gig. The circuit pulls less than 1mA from the battery, so battery will last many, many months.

Portable 9v Headphone Amplifier

 


High Quality One-IC unit, Low current consumption

After several requests by correspondents, the decision of designing a 9V powered Headphone Amplifier was finally taken. The main requirement was to power the circuit by means of a common, PP3 (transistor radio) alkaline battery. So, implementing a low current drawing circuit was absolutely necessary, though preserving a High Quality performance


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Portable 9v Headphone Amplifier

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Parts:

P1 = 22K
R1 = 18K
R2 = 68K
R3 = 68K
R4 = 68K
R5 = 18K
R6 = 68K
C1 = 4.7uF-25v
C2 = 4.7uF-25v
C3 = 22pF
C4 = 220uF-25v
C5 = 220uF-25v
C6 = 4.7uF-25v
C7 = 22pF
C8 = 220uF-25v
J1 = 3.5mm Stereo Jack
B1 = 9V Alkaline Battery
IC1 = NE5532-34
SW1 = SPST Toggle Switch


More:

• The appearance of the 5534 low-noise op-amp at a reasonable price was much appreciated by audio designers. It is now difficult or impossible to design a discrete stage that has the performance of the 5534 without quite unacceptable complexity.
• 5534 op-amps are now available from several sources, in a conventional 8-pin d.i.l. format. This version is internally compensated for gains of three or more, but requires a small external capacitor (5-15pF) for unity-gain stability. The 5532 is a very convenient package of two 5534s in one 8-pin devices with internal unity-gain compensation, as there are no spare pins.
• The 5534/2 is a low-distortion, low-noise device, having also the ability to drive low-impedance loads to a full voltage swing while maintaining low distortion. Furthermore, it is fully output short-circuit proof. Therefore, this circuit was implemented with a single 5532 chip forming a pair of stereo, inverting amplifiers, having an ac gain of about 3.5 and capable of delivering up to 3.6V peak-to-peak into a 32 Ohm load (corresponding to 50mW RMS) at less than 0.025% total harmonic distortion (1kHz & 10kHz).

Power Meter Schematic For Audio Amplifier

This is a simple schematic of audio power meter using LM3915 IC, this schematic can be used to measure the actual output power of your amplifier. For an audio engineer, this schematic Seems to be very helpful, especially for checking of sound system installation and field testing. Due to its logarithmic scale, the wide range of audio output with only ten scales cans Also be measured. If you take an attention to the pin number 5 of the LM3915, see That you will from the input is not yet Rectified. At this input pin, the negative swing will from the present. However, it is harmless since the current is limited by R1, the LM3915 earnest therefore respond only to positive cycle.

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 Power Meter Schematic For Audio Amplifier

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Note:

When the speaker resistance is 4Ω, then, make R1=10kΩ, if the resistance of speaker is 8Ω, make R1=8kΩ, and if the resistance of speaker is 16Ω, make R1=30kΩ.

The absence of peak detector or the detector will of averages give the circuit a fast reading of instantenous power, and this Gives us insight of both average and peak condition. For more readable peak or average mesurement, you cans use peak or average detector circuit.

Sound Level (Decibel) Meter Circuit

This is a decibel meter electronic circuit, For an audio engineer, this circuit seems to be very helpful, especially for checking of sound pressure levels from about 60 to 70 Decibel (dB). EACH light represents about a 3dB change in sound level so That Pls all three lights are on, the sound level is about 4 times Greater than the level needed to light one lamp. The sensitivity cans be adjusted with the 500K pot so That one lamp comes on with a reference sound level. The other two lamps will from then indicate about a 2X and 4X increase is in volume.

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Sound Level (Decibel) Meter Circuit

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In operation, with no input, the DC voltage at pins 1,2 and 3 of the op-amp will be about 4 volts, and the voltage on the (+) inputs to the 3 comparators (pins 5,10,12) will be about a half volt less due to the 1N914 diode drop. The voltage on the (-) comparator inputs will be around 5.1 and 6.5 which is set by the 560 and 750 ohm resistors.

When an audio signal is present, the 10uF capasitor connected to the diode will charge toward the peak audio level at the op-amp output at pin 1. As the volume increases, the DC voltage on the capacitor and also (+) comparator inputs will increase and the lamp will turn on when the (+) input goes above the (-) input. As the volume decreases, the capacitor discharges through the parallel 100K resistor and the lamps go out. You can change the response time with a larger or smaller capacitor.

Rangkaian Speaker Protector sederhana

Speaker Protector sederhana

This circuit follows will connects the speakers to the Power amplifier output only a few seconds after the amplifier is powered ON, so that the speakers do not accept popped up by a high voltage and you would not Hear a loud thud sound from the speakers When the amplifier is switched on. This stuff is very harmful to the speakers.

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Rangkaian Speaker Protector sederhana 1
Rangkaian Speaker Protector sederhana 2

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When the amplifier is powered on the bridge D1 also gets powered through the amplifier’s power switch. Capacitor C1 filters the output of bridge rectifier D1. When the power switch is made ON, the transistor Q1 gets switched ON only after the capacitor C2 is sufficiently charged (0.7V) through the resistor R1. Here the value of C2 and R1 are so selected that the time delay is around 2 seconds. So the relay gets activated only after a few seconds the amplifier is powered ON and until that time the speaker will be kept isolated from the amplifier’s audio output as the speaker is connected to the amplifier’s output through the N/O contact of the relay. During this initial delay period the output of amplifier will be grounded by the resistor R2 through the N/C contact of the relay. This is done in order to ensure that the DC blocking capacitor at the amplifier’s output is charged before it is connected to the speaker.

Rangkaian Relay Penunda Speaker

The following is a series of speakers delay relay useful as a delaying the speakers. The purpose of this delay circuit is as a deterrent to prevent pounding of the very first amplifier is turned on.

This delay circuit has the advantage that is: if there konsleting, relationship with the speaker amplifier circuit will be immediately cut off, so that the speaker is safe from fire. waiting time or delay time is about 5 seconds, and can be altered by changing the condenser C2 with elco smaller value. The time needed to re-open relay if the current cut off about 0.5 seconds.

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Rangkaian Relay Penunda Speaker

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About Relay Contacts


A relay is an electrically operated switch. Electric current through the coil of the relay creates a magnetic field which attracts a lever and changes the switch contacts. The coil current can be on or off so relays have two switch positions

Gambar Relay PWR SPST 16A 24VDC

A simple electromagnetic relay, such as the one taken from a car in the first picture, is an adaptation of an electromagnet. It consists of a coil of wire surrounding a soft iron core, an iron yoke, which provides a low reluctance path for magnetic flux, a movable iron armature, and a set, or sets, of contacts; two in the relay pictured. The armature is hinged to the yoke and mechanically linked to a moving contact or contacts. It is held in place by a spring so that when the relay is de-energised there is an air gap in the magnetic circuit. In this condition, one of the two sets of contacts in the relay pictured is closed, and the other set is open. Other relays may have more or fewer sets of contacts depending on their function. The relay in the picture also has a wire connecting the armature to the yoke. This ensures continuity of the circuit between the moving contacts on the armature, and the circuit track on the Printed Circuit Board (PCB) via the yoke, which is soldered to the PCB.

3VDC Stereo Headphone Amplifier by IC LM4910

 

This is a stereo headphone amplifier circuit of using LM4910 IC. This circuit is very simple so it is very easy to make even for beginners once. because the voltage source is used only 3 VDC you can use 2 x A3 battery in series stacking.

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3VDC Stereo Headphone Amplifier by IC LM4910

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Circuit diagram of 3VDC Stereo Headphone 

Amplifier by LM4910 

IC LM4910 Pinout

LM4910 Boomer belonging to the series of National Semiconductors is an integrated stereo amplifier is primarily intended for headphone stereoapplications. The IC can be operated from 3.3V ans its 0.35mW can deliver output power into a 32 ohm load. The LM4910 has very low distortion (less than 1%) and the shutdown current is less than 1uA. This low shutdown current makes it Suitable for battery operated applications.

C1 and C2 are the input DC decoupling capacitors for the left and right input channels. R1 and R2 are the respective input resistors. R3 is the feed back resistor for left channel while R4 is the feed back resistor for the right channel. C3 is the power supply filter capacitor. The feedback resistors also sets the closed loop gain in conjunction with the corresponding input resistors.

Rangkaian 11-90 hz Subwoofer Filter Using TL072 Op-Amp

The subwoofer filter circuit allows the addition of subwoofers to an existing full-range system, offering adjustable low-pass filtering with optional R6 and R8 boost and mono-summing.

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Rangkaian 11-90 hz Subwoofer Filter Using TL072 Op-Amp

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Skema Rangkaian 11-90 hz Subwoofer Filter
Using TL072 Op-Amp

TL072 Op-amp

The Subwoofer filter circuit to remove for separate pre amplifier to drive the low frequency sound a lot. In tone, call tone, normal Can not be done … is a fine deep low bass sounds like a bass drum, or at a movie complex in a low voice if we can be heard with But to add cabinets and amps. The subwoofer circuit is pass low frequency with in 11-90 Hz. Switching power supply 12V cut out if they need to use +-15V. I had change the Capacitor to cut out vocals per the red circle mark.

Audio Peak Level Indicator By Op-Amp

simple circuit PEAK indicator of foliage of the musical signal. This circuit was designed to provide a valuable test equipment tool for sound reinforcement systems like sound amplifiers and the like. The circuit is formed by an input buffer and ac to dc voltage converter (IC1A) feeding a window comparator (IC2A, IC2B, IC2C) which illuminates one of three LEDs at a time.

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Audio Peak Level Indicator By Op-Amp

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No setup is required: if correct values are used for resistors R3 to R7, LED D1 will illuminate at 0dB input (0.775V RMS), LED D2 at +5dB input (1.378V RMS) and LED D3 at +10dB (2.451V RMS).

The circuit was optimized for low current consumption as it was intended for battery operation. To achieve this, the best arrangement has proven to be the one using two different op-amp types for IC1 and IC2. In fact the LM393 IC was not operating satisfactorily as dot-mode LED driver, whereas the LM324 was unable to charge C2 in the linear way, as expected. Therefore, the final circuit is some op-amp wasting, but the small added cost will be quickly compensated by battery savings.

List Component:

R1    : 300K         D1,D2,D3 : LEDs
R2    : 1M2          IC1      : LM393
R3    : 510K         IC2      : LM324
R4    : 220K         IC3      : 78L05
R5    : 91K          SW1      : SPST  Toggle or Slider Switch
R6    : 160K         B1: 9V  PP3 Battery
R7    : 56K
R8,R9 : 100R
R10   : 220R
C1    : 100nF
C2    : 1µF/63V
C3    : 10µF/25V

Rangkaian Audio Surround Decoder

This circuit has been created to design a decoder circuit that will operate in a module that would produce an audio surround sound.
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Rangkaian Audio Surround Decoder

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Skema Rangkaian Audio Surround sound Decoder

The operation of the above circuits starts as the stereo sound signal transports surround sound information on the master volume part of the circuit. This will drive the Left channel Lch attached to Model TL072 IC1A and IC1b in which Right channel Rch is attached. The outputs on these operational amplifiers would serve as the input buffer to the following stages of the circuit. IC2C is responsible for summing up the signals from the left and right channels that will power the central loudspeaker output while IC2D is responsible for increasing the phase difference between left and right channels which is encoded in the two channels and will be fed to the rear loudspeakers. It is necessary to ensure that the negative terminals between the rear speaker is not earthed because they will simply function in parallel with the main speakers.

The output of IC2D will power regulated delay unit of audio to the rear loudspeakers. This would lead to the creation of proper sense of spacing in accordance to the size of the room. This will incorporate op-amp sound delay signal IC5 MN3004 which has 512 stages. Since IC4 MN3101 is a clocking signal, it provides timing to IC5 as it functions as an oscillator in the circuit. Variable capacitor C17 regulates the delay time in the circuit. The presence of filters in the circuit is for the purpose of preventing noise that will be produced during the process. These filters can be regulated to cut the frequencies above 8 KHz and under 100 Hz, to be able to drive the rear speaker. The rear loudspeaker is small in size because its input is encoded with a bandwidth of 100 Hz up to 8 KHz. The filters are built around the IC6A/B which is also an output buffer. A potentiometer is placed in every output to aid in the adjustment and regulation of loudspeakers and amplifiers. The supplied power in the circuit is 15 V and every output can drive a single power amplifier.

List Compoment

R1-2-7-8-12-13-18-19-20 : 47Kohm
R3-4-5-6-21-22-34-35    : 10Kohm
R9-10-11-14-15-16-17    : 15Kohm
R23-24-25-33-36         : 100ohm
R26-27-28-31-32         : 100Kohm
R29-30                  : 5.6Kohm
C1-8                    : 47uF/25V
C2-7-9-14-23            : 47nF
C3-6                    : 1uF/100V
C4-5-10                 : 33pF
C11-12-15               : 10uF/25V
C13                     : 82nF
C16                     : 18pF
C17                     : 100pF mini adjustable capacitor
C18                     : 2.2nF
C19                     : 4.7uF/25V
C20                     : 100nF
C21                     : 10nF
C22                     : 180pF
C24                     : 150nF
RV1-RV2                 : 2 X 10Kohm  Log. pot.
RV3-4                   : 10K Log pot.
D1                      : 1N4148
IC1-6                   : TL072
IC2-3                   : TL074
IC4                     : MN3101
IC5                     : MN3004 

Op-Amp 6-Line Audio Mixer Circuit

 

As with any audio mixer circuit, a slight loss is always introduced. The final summing amplifier has a gain of 2 or 6dB to overcome this. The Input line level should be around 200mV RMS.
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Op-Amp 6-Line Audio Mixer Circuit

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The mic inputs are amplified about 100 times or 40dB, the total gain of the mixer including the summing amplifier is 46dB. The mic input is designed for microphones with outputs of about 2mV RMS at 1 meter. Most dynamic microphones meet this standard.

The choice of IC op-amp is not critical in this circuit. Bipolar, FET input or MOS type op-amps can therefore be used; i.e 741, LF351, TL061, TL071, CA3140 etc. The power supply is a dual positive and negative supply, two 9 Volt batteries may be used as shown above or a power supply is recommended for longer periods of use

Rangkaian Alat Bantu Pendengaran

This circuit, connected to 32 Ohm impedance mini-earphones, can detect very remote sounds. Useful for theatre, cinema and lecture goers: every word will be clearly heard. You can also listen to your television set at a very low volume, avoiding to bother relatives and neighbors. Even if you have a faultless hearing, you may discover unexpected sounds using this device: a remote bird twittering will seem very close to you.

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Rangkaian Alat Bantu Pendengaran

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The heart of the circuit is a constant-volume control amplifier. All the signals picked-up by the microphone are amplified at a constant level of about 1 Volt peak to peak. In this manner very low amplitude audio signals are highly amplified and high amplitude ones are limited. This operation is accomplished by Q3, modifying the bias of Q1 (hence its AC gain) by means of R2.
A noteworthy feature of this circuit is 1.5V battery operation.

List Component

P1        : 22K   Log. Potentiometer
R1,R9     : 10K
R2        : 1M
R3        : 4K7
R4,R7     : 100K
R5        : 3K9
R6        : 1K5
R8        : 100R
C1,C2     : 100nF
C3,C6     : 1µF/63V
C4        : 10µF/25V
C5        : 470µF/25V
D1        : 1N4148
Q1,Q2,Q3  : BC547
Q4        : BC337
MIC1      : electret microphone
SW1       : SPST Switch
J1        : Stereo 3mm. Jack socket
B1        : 1.5V Battery (AA or AAA cell etc.)

9 Volt Portable Headphone Amplifier Circuit

Here I present a very simple and powerful headphone amplifier Circuit using NE5534/2 . In addition to the IC NE5534/2, the circuit uses only few passive components and can easily generate a lot of sound from even the most inefficient headphones and there will be no compromise for the quality.

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 9 Volt Portable Headphone Amplifier Circuit

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The 5534/2 is a low-distortion, low-noise device, having also the ability to drive low-impedance loads to a full voltage swing while maintaining low distortion. Furthermore, it is fully output short-circuit proof. Therefore, this circuit was implemented with a single 5532 chip forming a pair of stereo, inverting amplifiers, having an ac gain of about 3.5 and capable of delivering up to 3.6V peak-to-peak into a 32 Ohm load (corresponding to 50mW RMS) at less than 0.025% total harmonic distortion (1kHz & 10kHz).

List Component of Portable Headphone Amplifier

P1 = 22K
R1 = 18K
R2 = 68K
R3 = 68K
R4 = 68K
R5 = 18K
R6 = 68K
C1 = 4.7uF/25v
C2 = 4.7uF/25v
C3 = 22pF
C4 = 220uF/25v
C5 = 220uF/25v
C6 = 4.7uF/25v
C7 = 22pF
C8 = 220uF/25v
J1 = 3.5mm Stereo Jack
B1 = 9V Alkaline Battery
IC1 = NE5532 or NE5534
SW1 = SPST Toggle Switch

Circuit From: www.redcircuits.com

Megabass Circuit

The following is megabass circuit schematic (rangkaian megabass) . The megabass circuit is a modified Baxandall  tone control with no bass cut and no treble control. It boosts frequencies from about 30Hz to 160Hz can boost by 14dB.

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 Megabass Circuit

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Skema Rangkaian megabass

Note:

• The input capacitor can be replaced with a .01uf cap if you wish.
• The 10pf capacitor is optional and will start rolling off everything over 15kHz. 5pf will double this to 31kHz.
• The tone control requires a low impedence input. If you already have a low impedence input, the input buffer can be removed. However, the output is inverted.
• The opamp is not critical. A 4558 would be just fine.
• I do not show the parts for the +4.5 reference. Here is the +4.5 voltage divider I used.

IC A4558 Pinning

The A4558 is a monolithic Integrated Circuit designed for dual operational amplifier.

Absolute maximum ratings of A4558 Ap-amp

• Supply voltage VCC 20 or ±10 V
• Differential input voltage VIND 20 V
• Input voltage VIN ±10 V
• Power Dissipation PD 300 mW
• Operating temperature Topr -45 ~ +85 °C
• Storage temperature Tstg -55 ~ +150 °C

Audio Channel Selector

 

This circuit serves for connecting the stereo outputs from Four Different channels as inputs and only one of Them is selected to the output at any one time.

When the circuit switch on, channel A (AR and AL) is selected. If no audiois present in channel A, the circuit Waits for Some time and then Selects the next channel (channel B). This search operation continues Until it detects an audio signal in one of the channels. The inter-channel delay time or the wait Can be adjusted with the help of preset VR1. If still longer time is needed, May replace one capacitor C1 with of higher value.

To manually skip over from one active channel to another active channel, simply push the skip switch (S1), until the desired channel input gets selected. The selected channel (A, B, C, or D) is indicated by the glowing of corresponding LED (LED11, 12, 13, or 14 ).

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Audio Channel Selector

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IC CD4066 contains 4 analog switches, These switches are connected to four separate channels. These analogue switches are controlled by IC CD4017 outputs. CD4017 is a 10-bit ring counter IC. Since only one of its outputs is high at any instant, only one switch will be closed at a time. IC CD4017 is configured as a 4-bit ring counter by connecting the fifth output Q4 (pin 10) to the reset pin. Capacitor C5 in conjunction with resistor R6 forms a power-on-reset circuit for IC2, so that on initial switching on of the power supply, output Q0 (pin 3) is always high . The clock signal to CD4017 is provided by IC1 NE555 which acts as an astable multivibrator when transistor T1 is in cut- off state.

IC5 KA2281 is used here for not only indicating the audio levels of the selected stereo channel, but also for forward biasing transistor T1. As soon as a specific threshold audio level is detected in a selected channel, pin 7 and/or pin 10 of IC5 goes low . This low level is coupled to the base of transistor T1, through diode-resistor combination of D2-R1/D3-R22. As a result, transistor T1 conducts and causes output of IC1 to remain low as long as the selected channel output exceeds the preset audio threshold level.

Presets VR2 and VR3 have been included for adjustment of individual audio threshold levels of left and right stereo channels, as desired. Once the multivibrator action of IC1 is disabled, output of IC2 does not change further. Hence, searching through the channels continues until it receives an audio signal exceeding the preset threshold value. The skip switch S1 is used to skip a channel even if audio is present in the selected channel. The number of channels can be easily extended up to ten, by using additional 4066 ICs.