Friday, September 30, 2011

Surround Sound Decoder


The surround decoder is based on the "Hafler" principle, first discovered by David Hafler in the course of the 1970s. The original idea was to connect a pair of speakers as shown in Figure 1, for use as rear speakers in the surround setup.

It is fine as it is, but the problems are created when the main speakers are bi-amplification or transitional assistance, as it is a sign of full range output / total available for the rear speakers. There is no way to control the playback level, because it always ends up being the difference signal between left and right.

If the signal is mono, the signal on both channels is always more or less the same, and should not be starting from the rear speakers at all.
 The Original "Hafler" Surround-Sound Matrix

This circuit works by allowing the rear speakers to play only the difference signal between left and right outputs. All stereo encoded material has a certain difference between left and right (otherwise it would be mono), and this difference is that the output signal from the rear speakers.

It is important to ensure that the connection between the rear speaker terminals are unfounded negative, or they are simply in parallel with the main speakers.

Version line of passive level

So if you want to use separate amplifiers for the rear speakers, basically you can not - if you get smart. The first circuit shown in Figure 2 is completely passive, but requires that a suitable transformer. A suitable transformer means a line level, impedance of 10k units 1:1 - these are scarce, but are available after a search.

You may be able to get away with 600 units of ohms, but because the impedance needs, its performance is very common, with an extreme lack of bass (there are not enough inductance is 600 ohms transformer to operate at high impedance). Transformer is loaded to give back some of the low, but the preamp is likely to be very satisfied with the impedance. That said, I used this application to telecommunication transformers (600:600 ohm) and seem to work well.
Passive Line Level Hafler Matrix Decoder

The circuit is not a bad compromise, even if the impedances are too low to non-solid state preamp (preferably operational). Using a telephone adapter (600 Ohm), the loss is about 3 dB low frequency-3dB point around 100Hz. This varies depending on the quality of the transformer used, so experimentation is necessary. Although the 600 ohm telephone transformer are relatively easily available, some of them are quite common.

My tests were in a very good built by an Australian company called Transcap. I think I can say with some certainty that they will be reluctant to sell a single quantity. Another transformer manufacturer Midcom is very good in the U.S., but you will have the same problem with them. These manufacturers are prepared to deal with large orders from other companies, not people like you and I want ("Want a ...?") transformer. Therefore, you have to take everything you can get.

Since it is unlikely to be viable for most manufacturers, the alternative is to go active, using an ADC to perform the functions. This is described below.

The new circuit

The diagram in Figure 2 is a simple way to achieve the same (with some additional benefits) are online (ie, before the signal reaches the power amplifiers - in a bi-amp, this circuit is to be among the preamplifier and electronic crossover). The extras available are obvious ...

- The wiring is simplified (even if the power amplifier required)
- Now we have a center channel signal is available
- Provision for a mono signal to a sub-woofer is easy
The Schematic of an Enhanced Hafler Matrix Decode

While there have been published in the same circles over the years, this is a bit 'different areas. I wanted to avoid all the active electronics of the main left and right channels, as it eliminates any possibility of sound degradation due to the introduction of the threshold. Input impedance 50k does not cause any problems with the preamplifier (including types of valves), and the most important signal is simply a parallel circuit with the extras.

The volume is not included, because you already have a pre-amplifier. It would just become another component of the violin, and it would be little used, probably would have become a noisy time just sitting in a permanent position.

How it works

Opamp U1A is connected as an amplifier of the subtraction. If the same signal is used both as input, the output is zero. Consequently, it will remove all the background information from the stereo signal, and reproduce only the difference signal - exactly the same way as the original design Hafler.

U1B is a simple sum of the amplifier, and the food contains all the information as well as left and right channels. Ability to mind is that you could reduce the difference between the proceeds of this information, so the only material that is completely shared, both channels are reproduced. This would improve performance to the extent that an additional circuit is justified? I tend to doubt it, but you can pursue the matter.

Central control of the channel

Pot (VR1) is to set the center channel. This can be a TRIMPOT, or mounted on the back of a traditional dish (to help prevent the "fiddlers" from mucking your settings). I have seen circuits that do not have this, which basically feels like a bad idea. When two channels are added, the center channel is usually a level 3 dB compared to the left and right channels - if the signal is mono. Speech center channel (for example) is mono, so the standard is the same for each of the main speakers. Because the center channel amplifier and speakers are rarely as powerful as the left and right channels, it is very possible, the congestion amplifier, speakers, or both.

From the center channel is only supposed to fill "gaps" and provide a stable center of the image, which need not be so hard - especially because it is almost certainly lower than the main speakers sound quality and therefore reduce the overall sound quality. Level control lets you set the level just sufficient to provide a stable sound image, and no more. On my system, did not use a channel, and would have a negative impact on sound quality. If you have a good main speakers and a sound image stable and well defined, a center speaker can do more harm than good.

The capacitor (C1) is optional. Provides a nominal value of 8 kHz frequency roll-off (which apparently is quite normal for "real" surround sound processor). This helps minimize interference with the main stereo signal, but feel free to ignore, like most center channel speakers probably will not be able to play above that frequency anyway.

Subwoofer Out

The subwoofer output is simply taken directly from the mixing of the central canal, and I do low-pass filter, because I do not know any of the subgroups that do not have a filter in advance. Add another simply adds unnecessary complexity and will introduce the phase shift at the output of a phase compensation circuit (often included in the sub-woofer) may not be able to cope.


100 ohms at the exits to prevent the ability of the signal carried by the amplifier to oscillate. At that value, will not cause high frequency loss, unless you insist on 100 carries the signal of time (in my experience, these are rare).

It will also be noticed that there are two outputs for the rear speakers, just parallel. I included it because it is easier to thread if the user connects a stereo amplifier for rear speakers. Naturally, a mono amp do just fine as it is capable of performing the two rear speakers in parallel. It may not be possible if the speakers are 4 types of Ohm (it is becoming increasingly common in Hi-Fi, so it's not so stupid).

Tuesday, September 27, 2011

Sub Woofer Filter

The sound spectrum spans 20Iz very low frequencies and reaches the 20000Iz at high frequencies. At low frequencies is degraded in the sense of direction. This reason leads us to use the loudspeaker for the allocation of very low frequencies. Manufacture of you, we suggest to distinguish between these frequencies lead to the corresponding amplifier. Acoustic filters are located in various parts of sound systems. Baxandal knownest The application filters to control the frequencies of light tone and low and high acoustic filters, which divides the region into sub-areas, leading to the speakers. The application you propose is a simple filter in the region that limits the acoustic region (20-20000Hz) in the region of 20-100Hz.



Parts Layout

R1 = 39 Kohm
R2 = 39 Kohm
R3 = 47 Kohm
R4 = 10 Ohm
R5 = 22 Kohm
R6 = 4,7 Kohm
R7 = 22 Kohm
R8 = 4,7 Kohm
R9 = 10 Ohm
R10 = 220 Ohm
C1 = 39 pF
C2 = 0.1 uF
C3 = 0.1 uF
C4 = 0.2 uF
C5 = 0.4 uF
C6 = 0.1 uF
C7 = 0.1 uF
IC1 = TL062/TL072/TL082

With a production that aims to make an active filter, in order to bring a very low frequency speaker. This can put a speaker among the largest hi-fi speakers. To give you an overview of audio you also need the equivalent of an amplifier. The input circuit that is to connect the two outputs, and the preamplifier or the output line of some of the preamplifier. Production circuit afford to leave to lead a subwoofer to the power circuit. If for some reason you do not have space, so that you can enter the third speaker in the listening space, you can choose a smaller speaker. Power depends on the type of music you hear. If you have room, then when you make a permanent filter, and thanked, it may recommend to friends, or even do nothing, while your friends.

Audio Processor Circuit for Electronic Music Applications

This tour offers an audio processor IC SSM2045, specially designed for use in electronic music and the operational amplifier IC developed circuit 741 is configured as a low pass filter with a DC control voltage to win. The input signal is at a level of work set by the resistor R1 150mVpp.

How does the audio processor. The filter has two output buffers: output pin 2-1 pin and 4-pin 8-pin output. Internally, the outputs of two voltage-controlled amplifier (VCA) is connected.

R15 and R16 are connected to these outputs to obtain a change in the optimal control and eliminate tensions.

P4 is a volume control. Power to pin 15 and 16 shall not exceed a maximum of 250 uA. Balance between the two VCA, and the filter is either a full voltage range -250 mV + 250 mV in 14-pin control.

This tension can be adjusted P2.

Input can be controlled from up to 200 Ω source impedance. When the input level is 0 dBm, the VAC decreases by 6 dB. Bias current required to pin 17 is 120-185 Pa Pa. The cutoff frequency of 20 Hz to 20 kHz can be transferred to a variable voltage pin 5 This can be changed P1. Capacitance values ​​were chosen to spread the characteristics of the Butterworth filter.

The output current of the SSM2045 IC is converted into an output voltage of amplifier 741 All circuits must be disconnected from the DC sybsequent IC2. Voltage noise ration over 80 dB.

Saturday, September 24, 2011

Boss CE-2 chorus pedal

The Boss CE-2 chorus pedal is a good analogue of the late 70s and early 80s. According to this page on was the marketing of this 2-arrested in November 1982, but the pedal is still produced and sold until at least 1990. Although no longer in production there are thousands of these pedals used by guitarists around the world - it is one of the most coveted Boss pedals on the market opportunity.

Schematic :

Sunday, September 18, 2011

Audio Video Transmitter

The above diagram of a transmitter circuit television. Video input jack J1 first termination resistor R6 and capacitor C1 connected via clamping diode D1. Clamping forces sync pulses to a fixed DC level to reduce the effect of flowering. Potentiometer R3 is used to set the video signal of winning, its effect is similar to adjust the contrast of the TV. R7 bias can be adjusted to control the image of the black level, so that at a certain level, the signal is sent, the image even in total darkness. This TV receiver can maintain a proper synchronization. When we arrive, the potentiometers R3 and R7 are adjusted for cross-best all-around performance. RF transformer T1 and its internal capacitor form a tank Hartley oscillator circuit that is tuned to 4.5 MHz. Audio input signals to J2 is connected to the bottom of the C2-D3 and R4: the audio signal modulates the subcarrier signal that Q3 provides audio, s 4

5 MHz above the video carrier frequency. FM-modulated subcarrier applied to the modulator section C5 and R9. Resistance R9 adjusts the level of the subcarrier from the video signal. The transistors Q1 and amplitude modulated video and audio signals Q2 on a RF carrier signal. The primary frequency is set by the coil L4, which is 3.5 turns 24 - enameled wire on a form containing a ferrite coil slug.That standards is part of a Colpitts tank circuit also contains C7 and C9. The tank circuit forms of network so that responses Q4 Q4 turns on the RF output to adjust the frequency of the oscillator section is amplified by Q5 and Q6, the voltage from the modulator section. Adaptation of the antenna and low-pass filtering is performed by C12, C13, and L1. Resistor R12 is optional, it is added to help match the output of another antenna.


Before going further, while it is certainly possible to build the machine from scratch. But unless you're an experienced builder and hangers-qualified parts, it is strongly recommended to buy the full game, or at least, the kit component of the source included in the list part. While most parts are readily available, some can be a real headache to do.

The 4.5 MHz RF transformer (T1) used in the kit is an OEM Toko is not available through traditional sources. Although nearly all 4.5 MHz RF transformer is similar to that described in subsection (internal capacitor, hit secondary) can be used, these units are difficult to obtain an amateur friendly source. If you are determined to follow this path, it is best to contact Toko (1250 Feehanville Dr. Mt Prospect, IL 60056, Tel 708-297-0070 ..) for the location of your nearest dealer of the full range . In addition, the coil L4 is a unit of measurement. However, it can be done at home with the parameters listed above. The transmitter must be incorporated into a PC card for better performance. You can make a pension from the template sheet provided in Fig. 2, or use that comes with the kit. The parts are installed on the motherboard as shown in the chart location of the parties [see Fig. 3). Pay special attention to the orientation of the transistors, electrolytic capacitors. and the diode.

Outline the switch (S1), which is shown in Fig. 3 is the same with the switch, SPST button kit, which is normally open. You can use any switch to an alternative. A simple style fits the Board of Directors with a screw machine: stylus is ideal for many applications. Battery holder can be soldered onboard jumper or install a piece of wire or doublesided tape screws.When board is ready to be installed in the case. If available in the Ramsey Electronics installed allows the government down, and lift the top, yet harmonized. It also protects against the bottom edge of his shorts during alignment. You should check the solder side of the government thoroughly before installation, in the case.

D1—1N914 silicon diode
Q1-Q—2N3904 NPN transistor

(All fixed resistors are 1/4-watt, 5% units .)
R1, R2, R11—1000-ohm
R3, R7—1000-ohm trimmer potentiometer, PCmount
R4, R9, R10—10,000-ohm
R12—75-ohm (optional, see text)

C1, C8—100-mF, 16-WVDC, electrolytic
C2—2.2--mF, 50-WVDC, electrolytic
C3-C6, C11, C14, C15—001-mF, ceramic-disc
C7, C9—2.2-pF, ceramic-disc
C10—100-pF, ceramic-disc
C12, C13—68-pF, ceramic-disc

ANT1—Antenna, telescopic-whip
B1—9-volt battery
J1-J3—RCA jack, PC-mount
L1—0.15-mH miniature inductor
L2, L3—2.2-mH miniature inductor
L4—0.14- to 0.24-mH adjustable, slug-tuned coil
(see text)
S1—SPST, push-button switch, normally open
T1—4.5-MHz 1F-can-style RF transformer

Thursday, September 15, 2011

Portable Guitar Amplifier

Component Part

R1_____________22K 1/4W Resistor
C1_____________10µF 25V Electrolytic Capacitor
C2_____________100nF 63V Polyester or Ceramic Capacitor
C3_____________220µF 25V Electrolytic Capacitor
J1,J2___________6.3mm Stereo Jack sockets (switched)
SPKR__________8 Ohm Loudspeaker (See Notes)
B1_____________9V PP3 Battery or 3V Battery (2 x 1.5V AA, AAA Cells in series etc.)


For reasons of simplicity and robustness, the unit features a double bridge amplifier IC and a few others. For the same reason you can not control the volume or tone are provided as it is assumed that the existing controls on the electric guitar used successfully with the target.

No switch is used: the battery voltage is applied to the circuit when the jack is inserted into the input jack J1. To do this, make sure the input jack is a standard 1 / 4 inch mono guitar jack and J1 is a 1 / 4 inch stereo jack.

J2 output connector plug must be turned on the stereo type. The switch is arranged so that when a single stereo headphone jack is plugged in, the speaker will be disabled and the output disc at a time mono headsets in the series, giving full reproduction of the helmet. When used as an output Fuzz-box, a mono jack is connected to J2.

If the amplifier is to be enclosed in a pack of cigarettes, the speakers of standard diameter must be 57 or 50 mm.

Technical Data:

Max power: 1.5 W @ 9V - 8 ohms, 60mW @ 3V supply - 8 ohms

Flat frequency response from 20 Hz to 20 kHz

THD @ 100 mW: 0.2%

Max input voltage of 3V @: 8mV RMS

Minimum input voltage for the operation Fuzz-box: 18mV RMS @ 3V supply

Current consumption @ 9V 400 mW and power: 200mA

Current consumption @ 9V 250 mW and power: 150mA

Consumption 60mW @ 3V supply: 80mA

Standby current: 6mA @ 9V, 4mA @ 3V supply

Fuzz Box consumption: 3 mA @ 3V supply

Wednesday, September 14, 2011


Transistor amplifiers process the signals important. Many of them are driven so hard by large collector current input signal is either switched off or in saturation region for much of the input cycle. Thus, these amplifiers are generally classified by their mode of action. This classification is based on the amount of bias transistor and the amplitude of the input signal. It takes into account the part of the cycle, the transistor conduct. They are classified as

1. Class A amplifier.
In this case, the transistor is biased so that the flow of output current for the entire cycle of the input signal. Therefore, the operating point is chosen so that the transistor operates only in the linear region of its load line. As an amplifier can amplify the input signal of low amplitude. Since the transistor operating in the linear part of the load line, the output is exactly the same input. Thus, Class A amplifiers are characterized by a high fidelity output. These amplifiers are used when no distortion is the main objective. The operation is limited to a small central region of the load line for these amplifiers can be used to amplify may be used to amplify low amplitude signals. Also AC power transistor output is small. The global maximum efficiency, with a resistive load is 25%. The maximum possible performance of the sensor with a resistive load is 50%. If the power transformer is used, and the efficiency is 50%.

2. Class B Amplifier 
In this case, the bias transistor and amplitude are such that the output current flows only in the positive half cycle of the input signal. Zero signal collector current is zero and the weighting system is required for class B amplifiers Operating point is selected as a collector voltage cut-off. Given that the complete absence of negative side-cycle distortion of the output signal is high. Zero input signal is the best prerequisite for a Class B amplifiers because the zero IC. Transistor dissipates more power increases the signal strength. Compared to the average class of power amplifiers is less, the power loss is smaller. Thus, the overall efficiency has increased. The theoretical efficiency of the operation of class B is about 78.5%, compared with only 50% of the class action.

Class C Amplifiers
In this case, the bias transistor and the signal amplitude such that the flow of output current for substantially less than half of cycle-cycle entry. In such an amplifier is the foundation provided some negative stereotypes, so that the IC does not flow only when the positive half cycle of the signal begins. The effectiveness of the circuit is high (about 85-90%). Due to the high distortion, these amps are not used for audio-frequency. They are employed by radio frequency output power as high harmonic distortion can be removed by simple circuits.

4. Class AB Amplifiers. 
The characteristics of such an amplifier located between those of class A and B power amplifiers in class. The output current flows for more than half but less than the entire cycle.

Head Sinks 
Since the power transistors to handle large flows, they become very hot during operation and the heat generated is dissipated to the environment so that the temperature does not exceed the permitted limits. The transistor is usually connected to the metal plate (usually aluminum) to transfer heat to an aluminum plate. This is called a flat aluminum heatsink.

Friday, September 2, 2011

Scott Swartz PT-80 Delay

Component Layout