Hi-Fi Audio Amplifiers with TL072 Preamp Circuit

The amplifier is very easy to get on the board, and an innovative tone defeat. Instead of completely disabling the sound system, the massive de-sensitized, and they defeated a maximum range. This can be increased if desired, you can have two tone controls, the lifting of 10 dB and a cut and the other with a little help and very subtle 3dB cut - it is quite (surprisingly) a very small adjustment to how you need for day-to-day hearing.

Otherwise, the design is fairly standard, with a great advantage over other models that require virtually no cable. Download the source you want - I suggest you put a switch on the back of the cabinet, and an extension of the tree with the tree in front. This leads to a minimum of wiring and reduces the crosstalk from other active inputs.

Description of the circuit

The input stage is shown with a gain of 2 times (-6 dB), configure, and also acts as a buffer for the tone control. The tone control is a basic form Baxandall, but the addition of R117, 118 and 119 provides the flexibility and ease of reconfiguration, which is the traditional layout.

I have not seen before, this technique is used). As it is 100k, which limits the number of tonal control at a reasonable price + /-10dB. To increase further in section, R119 (R219 and receive) can be omitted. Conversely, reducing the value of a small area is also about 6 dB at 20 Hz and 20 kHz with 7.5dB at 22K.

Audio system (and overall) performance is shown in Figure 2 (in increments of 10% of the pot), and it seems that the midrange is not affected. This is contrary to most of the drawings, in which the control is aligned to 1 kHz, and is very audible in the media. For those who want to use the tone controls, I would suggest that both, with no tone controls are designed and in harmony with reality, minimalist design.

Low Voltage Mini Stereo Amplifier

The circuit is easy to build, it can be done easily on the edge strip, although I never tried to be honest. As always make sure that you insert TDA2822M chip on the board the right way and you get electrolytic capacitors in the right direction, I recently blew one of the greatest speakers I mentioned above. Nothing else is particularly critical, as the amplifier to operate with the lowest resistance and capacitors, you can buy. A log-log pot is recommended because it allows easy adjustment of volume. Everything else can be added as you need to know the tone controls, etc., but it is unlikely that you'll need something else for simple systems.

Voltage range betwen 1.8 V - 18 V

BOSS Octave OC2 Schematic

The OC-2 reproduces three different tones: the original tone, one octave below, and two octaves below. Each part has its own volume control so they can be mixed at will.

Simple Stereo Synthesizer

There are two common methods for generating a pseudo-stereo effect from a mono signal, the mono signal to both speakers play out of phase, and the use of frequency selective techniques, which usually consists of directing lower frequency signals in a single channel and higher frequency signals in the other. This circuit uses the second technique, but it can also antiphase signals that can give a better effect, especially when using headphones.

Q1 is used an emitter follower buffer stage, which provides two filter networks quagmire of its production is driven low impedance source. If they were driven directly to the input, it is quite possible that they would receive food source impedance of a few ohms or more pounds, which would be more than enough to change their properties effectively.

Both filters are formed by R4 and C3 (low pass), and C6 and R8 (step height). Wind in the price is not essential in this application and the rate of attenuation of 6 dB per octave filters simple as that RC is perfectly adequate. The-3dB point of each filter is about 800 Hz and mixed, therefore, gives an almost flat response, with significant peaks or troughs.

Q2 is connected as an emitter follower buffer stage, which ensures that there is minimum load on the low-pass filter. Q3 also ensures that there is minimum load on the high-pass filter, but this device can also serve as a phase separator. With SW2 switched to the output of the transmitter third quarter, Q3 acts effectively as an emitter follower and makes no phase inversion. With SW2 switched to exit in the third quarter of collectors, Q3, thus effectively act as a common-emitter stage with negative feedback of 100% (and a unity voltage gain) due to R11. 1t also provides a phase change of 180 °, so the two output signals are in phase opposition. A phase conditions are necessary to give a good stereo image center and use of anti-phase signals tend to give an impression of greater separation channel.

Stereo recording in an orchestra, it is normal for the violins from the left channel, with cellos and basses in the right way. Therefore, the high frequency signals are routed to the left channel and low frequency signals are routed to the right channel so that the device gives a similar effect (although this will obviously work well with the outputs connected both ways ).

Simple Scratch and Rumble Filter

This is 12 dB per octave and the beginning of the add-on Rumble filter, which can be connected to "Tape" or the institution responsible for the amplifier.

And 'the usual quadratic filter circuit having a passive high pass filter formed by the ability of C3 and C4 series and parallel resistor R2 and R3 (the latter is also used to bias the transistor Q1 source follower). Passive filter of this type gives only very slowly to the original roll, and the final reduction was only 6 dB per octave. Boot Strapping resistance is then used to improve performance. Above the cutoff frequency, where profit is the district would otherwise drop slightly, R1 is the effect of strengthening the signal. Well below the cutoff frequency, the loss of C4 leads to the signal emitter of Q1 is significantly below the junction of C3 and C4. This leads to some of the signals at the intersection of C3 and is pushed through the R1 and R1 with C3 effectively forming a second network-pass filter. This eliminates the sluts, the number of the original film (in fact, is a small and insignificant in a maximum of about 05dB above the cutoff frequency) and accelerates the depreciation rate for a nominal 12 dB per octave.

The low pass filter works the same way that the high-pass, except, of course, the elements of R and C of the filter have been implemented to provide the correct action of the filter.

With the component values ​​specified for Rumble filter response drops below unity at about 45 Hz, where -6 D13 just above 30 Hz, then drops to a nominal 12 dB per octave. The filter response from zero to unity gain points around 6k5Hz time 6dB points at about 10 kHz, then drops to a nominal 12 dB per octave.

Stereo Infra Red Transmiter

 TSH 512 Pin Out

 

Stereo IR Transmiter Schematic 

Stereo Infra Red Receiver

 TSH511 Pin Out

 

Shematic of Stereo IR Receiver

Noise reduction class-D headphone driver amplifier

The NE58633 is a noise reduction stereo Class D Bridge-Tied Load (BTL) helmet driver amplifier. Each channel has a Class D driver's helmet BTL amplifier, a electret microphone preamplifier for low noise, noise reduction circuit and returned to the music amplifier input.

NE58633 is the battery voltage from 0.9 V to 1.7 V, the chip employs on-chip DC-DC boost converter and the internal reference voltage Vref which is filtered and country of origin of noise removal. It is a mute control, and plop, and then click a reduction circuit. The amplifier gain of the microphone and amplifier of the filter is fixed with external resistors. Differential architecture provides better noise immunity.

The NE58633 is able to drive through a 800 mVrms 16 Ω or 32 Ω load and offers Electrostatic discharge (ESD) protection and short circuit. Available in 32-pin HVQFN32 (5 mm x 5 mm x 0.85 mm) package for high density of small and layout is ideal for noise-reduction headsets and audio teaching aids.

Caracteristic of NE58633

- Low current consumption of 4.4 mA

- 0.9 V to 1.7 V battery operating voltage range

- 1 % THD+N at VO = 1 VM driving 16 Ω with a battery voltage of 1.5 V

- 10 % THD+N at 800 mVrms output voltage driving 16 Ω and 32 Ω loads with a battery voltage of 1.5 V

- Output noise voltage with noise reduction circuit typically 31 mVrms for Gv(cl) = 25 dB

- On-chip mute function

- Plop and click reduction circuitry

- Class-D BTL differential output configuration

- Electret microphone noise reduction polarization amplifier with external gain adjustment using resistors

- Music and filter amplifier with external gain adjustment using resistors

- DC-to-DC converter circuitry (3 V output) with 2.5 mA (typical) load current

- Internal voltage reference pinned out for noise decoupling

- Available in HVQFN32 package

Block Diagram

 

Pin Out

Schematic Aplication

Balanced Mic Preamp

The preamplifier is designed for use with dynamic (moving coil TM) microphone with an impedance of 200 Ω balanced and terminals. It's a pretty simple design, which can also be seen as an amplification stage based on a single instrument type NE5534 op amps. To get the most common mode rejection (CMR) in a balanced signal, the reasons for the division of the dividers (R1-R4-R5 and R2, respectively) to the inputs of operational amplifier must be identical. Since this can be difficult to achieve in practice, a preset potentiometer P1 is connected in series with R5. The preset allows the common mode rejection optimal.

The capacitor C1 prevents input voltage while the resistor R7 ensures the stability of the amplifier with capacitive loads.

Resistor R3 prevents the amplifier goes into oscillation when the input has been interrupted. If the microphone cable is reasonably long, R3 is not necessary, because the parasitic capacitance of the cable to ensure the stability of the amplifier. It should be noted, however, that because the R3 improve the> 70 dB CMR> 80 dB. Performance of the preamplifier is very good. THD + N (total harmonic distortion plus noise) is less than 0.1%, where the input signal is 1 mV and the impedance of 50 Ω current. Under the same conditions, the signal to noise ratio is -62.5 dB. When the values ​​of the components have been defined, the gain is 50 dB ('316). After careful adjustment P1 at 1 kHz, CMR, without R3, is 120 dB. The supply voltage is ± 15 V. The power amplifier pulls the voltage of about 5.5 mA. Note: the removal of power lines with L1, L2, C2-C5.

Lie Detector

Here is a simple lie detector that can be built in minutes, but can be incredibly useful when you want to know if someone is really telling you the truth. It is not as sophisticated as those that the pros use, but it works. It works by measuring skin resistance, which goes down when you are lying.

Parts :

R1    33K 1/4W Resistor

R2    5K Pot

R3    1.5K 1/4W Resistor

C1    1uF 16V Electrolytic Capacitor

Q1    2N3565 NPN Transistor

M1    0-1 mA Analog Meter

MISC1Case, Wire, Electrodes (See Nots)

Notes

- The electrodes can be alligator clips (although they can be painful), electrodes (like the kind they use in    hospital), or just the son and bands.

- You can use the circuit, connect the electrodes to the back of the hand subjects, approximately 1 cm intervals. Then adjust the meter reading 0 Applications. He knows the subject is lying when the meter changes.

1500 watt power amplifier

Digital Equalizer

The series is I do here is a series regulator Volume, Bass and Balance digital Trable (tone control). The core of this circuit is an output-type IC manufacturer Maxim MAX5406, while the IC is an audio processor that comes with the switch successfully interface for tone control setup above. The circuit diagram is as follows

 

The above scheme uses support components and this makes very little can be done with only a small matchbox, even if used all SMD components can be reduced in size to half that time I'll just give Simply design, following pic:

 

Indor Intercom

In this circuit, intercom, an 8-ohm is used for both the microphone and output device. Phase BC109C extends a common base mode, in which case a good voltage gain, while providing a low input impedance of the speaker. Self-DC bias is used allowing increases of transistors in progress. LM386 is used in inverting mode amplifier power to increase the voltage gain and drives 8 ohm. 10k potentiometer volume control works, and the overall benefit can be adjusted using the 5k in advance. Double-pole switch, turn the position of the speakers, so you tend to speak and another to hear. Manually operate the switch (from inside the house) allows two-way communication.

HiFi Pre-amp

This is a HiFi low noise pre amplifier schematic. Wide frequency range around 10 Hz to 100 kHz will be won by the amplifier for maximum sound quality.

Surround Sound Decoder

Presentation

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.

Miscellaneous

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).

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.

Schematic

PCB

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.

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 BossArea.com 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 :

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.

Construction

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.

PARTS LIST

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

RESISTORS
(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
R5—47,000-ohm
R6—75-ohm
R8—4700-ohm
R12—75-ohm (optional, see text)

CAPACITORS
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

ADDITIONAL PARTS AND MATERIALS
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

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
IC1____________TDA7052
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.)

Notes:

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

CLASSIFICATION Amplifier

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.

3. 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.

Scott Swartz PT-80 Delay

 

Component Layout

Schematic

Mosquito Repeller

Description: The following diagram shows the electronic mosquito repellent. Features: The piezoelectric buzzer converts the amplifier output of ultrasound can be heard by the insects, an amplifier consisting of four transistors, mosquitoes can be postponed by the use of frequencies in the ultrasonic (above 20 kHz ) wide transistor to amplify the sound. Component: IC, transistor, resistor, President, condenser

Tascam MP-GT1

Tascam MP-GT1 to maintain the honor of being the first MP3 player designed for musicians. It is essentially a mobile guitar trainer, which will be enough memory for up to 240 songs. You can learn new riffs slow down guitar parts, loops, and also remove some parts. The MP3 playback is achieved through a variable speed audition, slows down the speed without changing pitch. You can also step up or down the current song to the tune guitar. MP-GT1 has a built-in tuner and a metronome, a graphic LCD display, and rotate the data for navigation. Tascam MP-GT1 is available in Japan since the end of this month at a price reserved.

Cracker Box IPod Amplifier

1. Toggle switch, single pole single throw
2. 9V battery
3. 9V Battery connector
4. 0.047µF capacitor
5. 220µF capacitor (biggest)
6. 0.01µf capacitor
7. 100µf capacitor
8. Hookup wire, 2 0 or 22 gauge AWG solid core is best.
9. 5KO potentiometer (audio or log taper)
10. 25-ohm (25O) rheostat
11. LM386N audio amplifier
12. 8-pin DIP IC socket
13. Chicken head knobs (2)
14. Prototyping PC board
15. Speaker, 8 ohm impedance
16. 10O resistor
17. 1/4 mono phone jack

SIGMADSP AUDIO PROCESSORS

SigmaDSP ® digital audio processors allow the use of a fully programmable audio DSP that can be easily configured through the graphical development tool SigmaStudio ™. Latest SigmaDSP SigmaDSP products against the car audio and portable our best performance and power aware processors. The SigmaDSP ADAU1442, ADAU1445 and ADAU1446 combine a core of 172 MHz with an array of routing, including sample rate converters asynchronous and S / PDIF Rx / Tx. Routing matrix allows the connection of various digital sources operating at different sample rates to easily connect the audio processor.

The first low SigmaDSPs, ADAU1761 and ADAU1781, contains the same powerful SigmaDSP elsewhere, and is connected to a stereo ADC and DAC SNR performance to more than 100 dB. AD1940 and AD1941 role in the computing power of high I / O channel count. ADAU1701 and ADAU1702 include a complete analog I / O, Digital I / O and function of stand-alone system that offers full sound processing on a single chip. ADAU1401A ADAU1701 with similar functions, but is designed specifically for the automotive industry, as well as set a wide range of temperature.

The software tool Sigma graphical development studio, programming, development and optimization SigmaDSP audio processors. Familiar audio processing blocks can be interconnected in a pattern, and the compiler generates code DSP-ready and a control group of the surface for setting parameters and tuning. This tool allows DSP engineers with no experience writing code for easy implementation in a DSP design, but still strong enough to meet the demands of experienced designers DSP. sigma relations studio with two boards Analog Devices evaluation and design to provide complete production circuit in real-time control IC.

With flow charts DSP Development of the signal, including sigma studio includes more features to accelerate the design cycle of concept release product. Sigma Studio provides tools for intuitive mode control registers, the calculation of tables of filter coefficients, to visualize the size of the filter and phase responses, generates C header files, and sequencing of a series of controls to ease your transition from studio to sigma deployment of the system on your microcontroller.

Evaluation kits, including evaluation boards and tools are now available SigmaStudio full for all products SigmaDSP. Check the SigmaDSP FAQ for any questions you may have. If you answer no, please direct your request to or accessing the Forum sigmadsp@analog.com SigmaDSP SigmaDSP EngineerZone.

Audio Processor

This chip audio processor features SSM2045 IC is specially developed for applications in electronic music, and 741 IC OPAMP. The circuit is configured to low pass filter with gain control voltage. The input signal is set to work through the resistance level of 150mVpp R1.

The filter has two buffered outputs: the output at pin 2 to pins 1 and 4 output pins to pin 8th Internally, the outputs are connected to two voltage controlled amplifiers (VCA).

The R15 and R16 are connected to these outputs to achieve optimum shift control and elimination of tension.

P4 is the volume control. The current flowing to pin 15 and 16 should not go beyond the maximum of 250 uA. The remaining two OCV and the entire filter is controlled as a voltage range -250 mV to + 250 mV to pin 14th

This tension can be adjusted P2.

Input can be done by up to 200 Ω source impedance. When the input level of 0 dBm, the VAC decreases by 6 dB. Bias current required to pin 17 is between 120 and 185 uA uA. The cutoff frequency can be shifted from 20 Hz to 20 kHz with a connector 5 This variable can vary through the P1. Capacitance values ​​were chosen to give a Butterworth filter of its properties.

IC SSM2045 output power supply voltage 741 converted to AMP. All sybsequent DC is disconnected from the circuit IC2.
Noise voltage is about 80 dB

IK Multimedia iRig Mic

If you are an amateur or a professional and want to record the sound quality on your iPhone, iPod, iPhone, IK Multimedia has introduced the IRIG micro that can do just that. This hand-held microphone can be used to record anything anywhere, whether a musician to record a song or a journalist who wants to record an interview. It can also be used for podcasts - in fact, for any type of recording.

IPhone, iPod, and make the iPad has built-in microphone, but these do not produce great sound. IK Multimedia IRIG microphone to easily connect your device and you're ready to record good audio quality.

There are three levels to get the pass, which can be adjusted, depending on the type of sound you want. Although you can use it as a PDA, you can even put on a stand if you want a hands-free use. Thank you for your applications, you can add sounds to your songs, if you are recording tracks, for example.

Works with all audio applications, and provides real-time monitoring. As long as your expectations are realistic and you do not expect studio quality recording, you should be pretty happy with this microphone