Audio Schematics

Stereo audio distributor

2011-09-13T19:52:00.000-07:00

This is a stereo audio distributor 1 in / 3 out. The term has developed a bit much, given that it only get the basic diagram of a doc from one of the TL072 manufacturers.

However, It have adapted slightly to take advantage of editing a version for single power supply (single, for example a car battery 12 V) and another version for dual power supply(balanced, for example + / -12 V)

The general block can see that the structure of audio distributor is very simple.

A first stage, called AMP, can have a high input impedance and have a low output impedance. This first stage, whose primary purpose is to bring the gain even if it is possible (despite the name given AMP), followed by three output stages in parallel:AMP1, AMP2 and AMP3 This provides three separate outputs BF, able to attack each of the amplifiers, recorders, or other audio equipment.

The diagram shows two identical modules, for use in stereo. Of course you can use onlyhalf for use in mono. The following description is based on the first module consists of U1 (top module on the diagram)

Single or dual supply?
The scheme has been adapted to allow a single circuit board, using a single supply ordual power The choice is made by the introduction of two jumpers (per channel), according to the following configuration:

PSU simple: set up the jumpers SW1, SW4, SW5 and SW8, and do not wire thepower supply decoupling capacitors C6 to C9.
PSU dual set up the jumpers SW2, SW3, SW6 and SW7, and wire the power supplydecoupling capacitors C6 to C9.

In the first case, the two input terminals of the op amp U1: A is brought to a positive DC potential equal to half the supply voltage, through resistors R4 and R5. In the second case, the two input terminals of the op amp U1: A is brought to a zero potential, themass, still through resistors R4 and R5. Similarly, in the first case the negative supply terminal of the op amp is grounded, whereas in the latter case, it is brought to a negative potential.
Note that the potential + V "half power supply" is provided by the pair of resistors R2 and R3, equal to values, which form a resistive divider bridge The common point of the bridge, there is a good half of the power supply voltage. It is around this voltage we canconsider as a virtual ground, the input signals are centered, in order to receive the maximum dynamic range.
Decoupling capacitors for power supply: it should be placed as close as integrated circuits. If you are unable, make sure at least that the connections between these capacitors and integrated circuits are wide and generously covered with solder. More

Graphic equalizer based LA3607

2011-08-26T08:27:00.000-07:00

Have you collect the equalizer as well? No? Why? Too complicated schemes, too many elements, and complex printed circuit boards? Well, not so bad.

The following, I will tell you how you can build a very good equalizer with a minimal set of components that do not require any configuration, with mocking simple printed circuit board, but at the same time having a very decent options, fully worthy of the car amp or a small home industry. The basis of the equalizer is a dedicated IC LA3607, manufactured by nebezysvestnoy Sanyo, more precisely, its semiconductor division.

The characteristics of the equalizer as follows:

Supply voltage 9B
Current consumption of signal 7mA
Adjusting the number of bands 7
Depth adjustment + /-12dB
Conv. gain 0.8
Conv. harmonic distortion 0.02%

So, in the scheme, two components - a chip itself, and seven bandpass filters consisting of two capacitors and one resistor, which, strictly regulated and increase or decrease the selected frequency. If the selected capacitor regulated following bands: 60Hz, 150Hz, 400Hz, 1kHz (davit!), 2.5 kHz and 15kHz 6kGts.

Of course, they can be counted on other frequencies. And there is a very simple formula:

The resonant frequency is as follows:

Since R0 and R hide inside the chip and is 1.2 ohms and 68 ohms respectively, the formula can be simplified to the following:

Values in the formula can be substituted or microfarads and kiloomah, then we get the output of kilohertz, or Faraday, and ohms, then we hertz.

Well, the quality factor of each filter is as follows:

Given the above circumstances, we find:

Capacity in this formula can be substituted with anything - dimensionless figure of merit.

Nourish all that stuff you need from a stabilized source, which can collect, such as KREN9A. Oh, and I hope it is clear that the diagram shows a diagram of one channel, but if you have a few - two, for example, it is necessary to collect two of the same channel here.

Digital volume control DD1 К555ИЕ7

2011-08-23T01:08:00.000-07:00

The Digital volume control circuit has been designed to work with external wired remote has 16 levels of volume-driven volume up button and reset button. DD1 - counter displays 16-bit binary code to control the T shaped voltage divider. DD2 is used as a clock. DD3 inverts the signal from the counter. The regulatory element is made on resistors R6/R7, R8/R9 and transistor switches that are connected certain resistance to ground formed a T shaped voltage divider.

Circuit note :
1 - turn up the volume
2 - reset
3 - Clock output

Digital volume control Parts list :
R1..R5 1K
R6..R9 22K
R10 - R14 3.9K
R11 - R15 2.2K
R12 - R16 1K VT КТ3102
R13 - R17 510
R18..25 22K
C1 10mkF*16V
C2 47mkF*16V
C3,C4 100mkF*16V
DD1 К555ИЕ7
DD2,DD3 К555ЛН1
DA1 КР142ЕН5А

Simple Audio Digital Volume Control

2011-08-23T00:28:00.000-07:00

This Simple Audio Digital Volume Control scheme for all its simplicity allows you to adjust the volume with two buttons! Chip - DS1669 0.1 uF capacitor.

There is a volume control knob, which does not rotate, and has two positions - left and right. This Digital Volume Control is realy simple.

Digital volume control MAX5440

2011-08-22T23:08:00.000-07:00

Digital volume control MAX5440 makes this circuit does not require any special programming knowledge to work, because it works on the principle of "plug and play".

It is interesting to use the encoder to adjust the level (up / down) and a button to select the mode (the volume / balance / mute). There are 6 pieces of LED is quite informative to show the volume and balance. Above is a Digital volume control MAX5440 schematic diagram that is included in the documentation.

Since task was to explore the possibilities of the controller, rather than building a complete device, and it made no sense to complicate the board (eg, voltage regulators or display elements).

The Digital volume control work smootly, a uniform change in the volume and balance during rotation of the encoder, the correct ignition of LEDs according to the established level. Properly perfected pressing buttons «Mute» and «Mode». It is difficult to find fault with the good work. Control the sound without clicks.

USB sound card based PCM2702

2011-08-22T22:38:00.000-07:00

click to enlarge

Make a USB sound card is easy if you use a chip from the PCM2702 IC BURR BROWN / Texas Instruments in it, you can create a fully functional USB sound card. Sound Card gets power from the USB port and has one stereo output. You do not need to install any drivers for Windows XP and Vista, because they are already inside. It remains only to connect and listen.

This USB soundcard scheme works very well. The efficiency was tested on Windows 2000, XP and Vista. Drivers are in the system because the sound card can be used immediately, without any additional software.
There are new chips PCM2704, PCM2705 an even better solution with rich functionality.

Below are the pcb layout :

Passive tone control

2011-08-22T20:52:00.000-07:00

Discrete, passive tone control circuit above does not violate the integrity and depth of sound image of any High-End Complex.

Component selection should be with caution (a deviation from the nominal value not exceeding 0.5%) eliminates the mismatch on the stereo frequency and phase response at discrete points Treble control (left diagram) and LF (right) and small (but sufficient for High-End) depth adjustment Tone (6 dB in increments of 2 dB) allowed to circumvent the main disadvantage of passive regulators - a large (in some controllers to 30 dB) attenuation.

In this case, the signal is attenuated by only 7 dB, which does not however - a tangible deterioration in the dynamic range of the entire audio system. One additional switch Speech Cut allows us to weaken the signal by 5 dB and further to suppress low frequency components below 100 Hz by 3-5 dB. This is recommended for improved intelligibility when listening to speech software. For normal matching the output impedance of the source (preamp) should be less than 1 kW, and the input load resistance (amp) - at least 100K. Passive tone control

Mini Audio Mixer "Impulse MM-04"

2011-08-22T20:36:00.000-07:00

This is Mini Audio Mixer schematic diagram , it has 5 inputs and a separate jack for reverb.

The four main input channels are identical, have independently adjustable input sensitivity, tone, panning sound. The fifth entry - linear channel. Despite the cheap and primitive, the work Mini Audio Mixer "Impulse MM-04" was quite satisfactory

Spectrum analyzer based PIC18F4550. Part 2

2011-08-21T20:59:00.000-07:00

In the first part Spectrum analyzer based PIC18F4550. Part 1, we met with the scheme, with the appointment of individual nodes and elements, over the principles of operation.
Microcontroller software is written entirely in C language, and it can be conditionally divided into 4 parts.

1. ADC sample data (sampling)

Implementation of sampling analog-digital converter is the channel every 50 ms RA0. This gives us a sampling frequency of 20 kHz (20,000 times per second). For the Fourier transform is very important that the samples were uniform and accurate. With this purpose in the subroutine get data from ADC introduced a short time delay, which was calibrated using an oscilloscope connected to the control point W4 on the board. Square-wave period as measured in this reference point should be exactly 50 microseconds. ADC works with a full 10-bit resolution, and the results are shifted down to 512 to implement a "virtual earth" input signal. This means that the resulting data will be in the range from -512 to 512, as required for the mathematical transformation FFT.

The procedure for obtaining the samples will take a little time to 32 ms (64 × 50 microseconds) for each cycle.

2. 16-bit FFT

Subroutine, which implements the calculation algorithm FFT, was found on the Web so (link to the original code is in the source code at the end of the article). Math FFT is complex and difficult to understand. The code has been slightly simplified and adapted for the PIC18F4550. Since the microcontroller has a hardware multiplier 8 × 8 as part of its arithmetic and logic devices have been optimized computing, allowing the compiler to correctly use the features of the microcontroller.

3. The calculation of the absolute values

The result of calculations by fast Fourier transform algorithm - a complex number, which consist of real and imaginary parts of two arrays. To get the result in a meaningful form, we need to compute the absolute value of complex numbers that we get using the Pythagorean calculation for complex numbers. This means the square root of the numbers, so we implement fast integer computation SQRT (), as any floating point calculations considerably slow down the process.

FFT algorithm and the computation of the absolute values of complex numbers occupy approximately 70 ms for each cycle.

4. Update LCD display

LCD display with a resolution of 128 × 64 pixels should be updated as quickly as possible. For this purpose, a simple algorithm was used drawing diagrams in the form of columns, which requires a minimum number of teams display.

Two switches are installed on the motherboard allow you to change the display scale (1 ×, 8 ×) and display mode (linear, logarithmic). Sub-display service is on time 45 ms.

Approximate speed of the display of a spectrum analyzer 1 frame for 150 ms, ie we obtain as a result 6.5 frames per second (or approximately 10 frames per second without using the LCD). Users can increase performance by reducing the amount needed to analyze the frequency (this leads to a reduction in the time of receipt of samples and computation algorithm FFT) or by using a high-speed display. In addition, users can alter the device for use in conjunction with LEDs.

The limiting frequency for the Fourier transform (the Nyquist frequency, Nyquist frequency) - 10 kHz. The device analyzes the frequency range of 32 (Hz):

1: 312.5 - 625
2: 625 - 937.5
3: 937.5 - 1250
4: 1250 - 1562.5
5: 1562.5 - 1875
6: 1875 - 2187.5
7: 2187.5 - 2500
8: 2500 - 2812.5
9: 2812.5 - 3125
10: 3125 - 3437.5
11: 3437.5 - 3750
12: 3750 - 4062.5
13: 4062.5 - 4375
14: 4375 - 4687.5
15: 4687.5 - 5000
16: 5000 - 5312.5
17: 5312.5 - 5625
18: 5625 - 5937.5
19: 5937.5 - 6250
20: 6250 - 6562.5
21: 6562.5 - 6875
22: 6875 - 7187.5
23: 7187.5 - 7500
24: 7500 - 7812.5
25: 7812.5 - 8125
26: 8125 - 8437.5
27: 8437.5 - 8750
28: 8750 - 9062.5
29: 9062.5 - 9375
30: 9375 - 9687.5
31: 9687.5 - 10000
NOTE: Nyquist frequency - the frequency of the analog signal, it provides a correct conversion to digital form. In the Nyquist sampling theorem was proved that the sampling rate must be at least twice the highest frequency signal being processed, so you can restore the original signal - so, for sampling a voice signal having a bandwidth of 4 kHz, 8 kHz frequency is needed in Otherwise, the inevitable distortions when playing back the signal in digital form.

Downloads

The source code for the microcontroller (compiler HiTech C18) - download

Spectrum analyzer based PIC18F4550. Part 1

2011-08-21T20:23:00.000-07:00

The project demonstrates the implementation of the sound spectrum analyzer for 8-bit microcontroller PIC18F4550 manufactured by Microchip . Spectrum analysis is performed using an optimized fast Fourier transform algorithm (Fast Fourier Transformation, FFT), written entirely in C language. Data Visualization (spectrum) is carried out in real time on a graphical LCD display with a resolution of 128 × 64 pixels.

Key Features:

sampling rate 20 kHz;
frequency range 312 Hz - 10 kHz;
the display speed of 10 frames per second;
display resolution of 128 × 64 pixels;

To calculate the values of the fast Fourier transform algorithm in the range of sound frequencies necessary to prepare the signal for further processing microcontroller. PIC18F4550 has an integrated multi-channel ADC, which can be used to measure voltage in the range 0 V - 5 V with 10-bit resolution (0-1023).

The main characteristics of the microcontroller:

core PIC18, the operating frequency to 48 MHz;
32 KB Flash-memory programs;
2 MByte RAM;
one 8-bit timer, three 16-bit timers;
USB interface, SPI interface;
13-channel 10-bit ADC;
35 I / O lines for general use.

Audio signal from audio line-out is an alternating signal wave with an amplitude of about 1 V. The lower oscillogram clearly shows the sinusoidal tone frequency of 5 kHz (reference point on the chart W2 ).

If such a file directly to Segal ADC of the microcontroller, we get only a small range of input voltage (0 V - 0.5 V), ie we obtain the values of only the top of the wave and the calculation of FFT algorithm will be incorrect.

In order to get the correct calculations, we need over the original signal to do some action. First you need to amplify the signal in order to use the entire working range of the ADC of the microcontroller (0 V - 5 V). In the second stage, we need to shift the signal at 2.5 V ("virtual" 0) to the ADC can capture all the values of the analog signal. Thus, the microcontroller will be able to do sampling positive and negative half-wave signal.

To this end, the device used a simple on-chip amplifier LM386-1 - we solve the first condition - the signal gain. Nutrition amplifier is +5 V, and because of this, we perform the second condition - positive offset signal. With the same purpose is permissible to use rail-to-rail operational amplifier (op amp output signal c scale, equal to the voltage supply).

Oscillogram below shows the shape of the signal after the amplifier LM386-1 (control point on the chart W3 ).

The device has a stereo input J4, but with two resistors (R1, R2) par 10k done mixing the two signals. With the potentiometer R3 is possible to regulate the amplitude of the signal. The output signal from the amplifier passes through a simple RC filter is 10 kHz, and the resulting signal to the input of the ADC of the microcontroller.

RC filter 10 kHz in our case used as a filter of "smoothing" for the FFT, which can not correctly detect a signal with a frequency above 10 kHz. It is worth noting: RC filter - the filter is simple and very inefficient, but this type of filter was chosen because of ease of implementation, since required all two passive elements. Users can replace this part of the scheme, such as a filter using the operational amplifier.

To display the data using a graphical LCD display ATM12864D , one based on the controller and KS0108B KS0107B. The display connects directly to a microcontroller with 8-bit bus. In addition there are three LEDs, which are used in the test mode conversion, "music-light." Two buttons SW1 and SW2 are used to switch display modes. The second connector J5 is designed to implement the through channel (eg, headphones). J2 connector is designed to connect the programmer and in-circuit programming of the microcontroller. To supply schemes applied voltage regulator LM7805 .

The developed one-sided printed circuit board is designed for use and installation of components with pin terminals.

However, a schematic diagram of the device is not complicated and can collect it on the breadboard. Possible to use another microcontroller, software compatible with the PIC18F4550 (for example, PIC18F2550 ).

Schematic diagram and printed circuit board (format and expressSCH expressPCB) - download

In the second part of this article, we look at the highlights in the software of the microcontroller.

LEDs Signal strength indicator on the amplifier

2011-08-20T05:53:00.000-07:00

Does your audio amplifer have signal strength indicator? I've always envied the fancy amps with a graphical level meter signal on the LEDs, such that you can find a stereo system. This is your chance to design your own signal level meter, which you can connect to your audio amplifier and see how the signal.

The device is based on the chip LM3915. This monolithic integrated circuit from National Semiconductor is the driver for our device. At the 5 pin of the chip, we serve an analog voltage and the circuit operates 10 diodes in a logarithmic scale 3 dB / step.

Audio detector

2011-08-19T10:26:00.000-07:00

This scheme for the reception of audio signals with an adjustable noise compensator. It can be used to generate a weak radio signal, for example, PDA (Pocket PC)

Schematic of Audio Detector :

The circuit consists of:

Input impedance
signal amplifier
Filter
Custom comparator (Noise canceller)

How to make mp3 player ' Sakura Open Source' Part 2

2011-08-19T10:16:00.000-07:00

PCB Layout :

Source
Source code written in C. I use the PCM compiler for the average bit controller (PCM midrange compiler) from the CCS. It costs about a hundred dollars and produces very compact code, a complete set of built-in functions and data types. I am very pleased with them. Web site compiler developers - ccsinfo.com .

For people who do not need to modify the code is. hex files, ready for loading directly into the PIC controller. Do not forget to clear the check box ( fuses ) and Brownout WATCHDOG , it would mean - to prohibit ih.Esli you do not, the entire device will refuse to work. This is the most common mistake.

To date, all of 4 kilobytes of program memory is almost fully occupied. 256 bytes of flash ( EEPROM) memory is completely used to store tables of the function Shuffle. There are several non-critical functions that can be removed to free up some space in memory, such as buttons and sub-interface mode random playback. If the microchip company when neither be to release the 18-legged controller with 8 kilobytes of memory, I will definitely use it for this project. You can also make the code more efficient than the code I have made. If you do, let me know what you did.

Parts List

Number	Part number	Description
1	PIC 16 LF 88, SOIC package (in April 2006 at mouser they were the cheapest).	PIC microcontroller.
1	VS1011b, SOIC body . I think I use vs 1011 e is possible, but will require minor modifications.	Chip MP3 decoder.
1	SCDA 1 A 0901 (available from Mouser . com )	Slot for SD / MMC card
1		Low-current red LED, housing 1206
1	CP-3523SJCT-ND	Headphone jack.
1	ECS-129-18-4 12.96 MHZ or equivalent	crystal at 12.96 megahertz.
2.4	311-1154-1-ND	22pF capacitor, 1206 case.
1	MCP1700T3302ETTCT-ND	Stabilizer 3.3 volts, SOT23-3 package.
6	BC1300CT	0.1mkF capacitor, 0805 case.
3	PCE3093CT	Mini joystick (4 buttons and a "choice").
1	Not defined	5 pin plug 1 inch.
1	SBH-431A-ND	3 cell Ni - MH battery ( AAAworks well L ).
1		1M resistor 1206 chassis.
1		1K resistor 0805 chassis.
1	478-3026-1	from 10 to 15 uF tantalum capacitor.

How to make mp3 player ' Sakura Open Source' Part 1

2011-08-19T10:12:00.000-07:00

Having bought the parts for $ 30 and with a good amount of patience, it is possible to build a MP3 player, open source, and ample opportunities for modification. It can be modified so that will accept commands over a serial interface, can be part of an art project, used as a voice for your next smart talking robot friend, or filled with music, and used as is. Insert the card you want any size, up to the theoretical maximum size MMC. All code and diagrams shown here, are distributed free.

Printed circuit board and source code
.

Main features of the player: playback in random order, basic navigation in the tracks, volume control, support for FAT32, support for fragmented files, unlimited number of files in the root and high-quality playback. Files with a bitrate of 256 kbit / s are played without interruption. Supports files with different bit rate, maximum - 320 kbit / sec.

Wav -files are also supported, but they must be lesser quality than CD , as long as you do not put a crystal oscillator for PIC controller with greater frequency. In my test device is working, but use the PIC controller clocked at more than 8 MHz at a supply voltage of 3.3 volts is problematic. I'm still working on playback wav -files on this if you are having fun results, reproducing the wav -files, let me know.

To write data to a memory card, you need a reader / writer. In many computers, such devices are built. External USB models are available for sale at a price of about $ 10. Check the format of the file card system, it should be FAT32 instead of FAT16.

For normal playback, download MP -3 files to the card (make sure that all files have the extension ". MP 3 ") and insert it into the player. Turn on the player and it immediately begins playing the first track. Files are played in the order in which they were recorded on a memory card.

To switch to shuffle, hold down the "next track» (next track) when you turn on the player. He will remain in this mode until the next cycle off / on. Once all the songs will be played, the player goes to sleep until the next cycle off / on.

Caution for beginners! This project is an intermediate level. You'll need things like the PIC programmer, some knowledge of programming in C, as well as quite good but not great knowledge of electronics. The oscilloscope also does not hurt. Be careful, the details are very small.

If you're new, check out these books and web sites:

compiler that I used in the project - a compiler for the average bit controllers (midrange compiler)).

All three books Forrest Mims (Forrest Mims) are available in the radio room (Shop radioshak . com ).

Practical Electronics for Inventors, by Paul Scherz and journal Cheknuty and Volta (in the book of the fortress (online shop) they have almost always.)

Why is the project named Sakura? This project began when in bloom everywhere cherries. Sakura means cherry blossom in Japanese. And the first prototype of the project was housed in a pink plastic.
The scheme :

RIAA equalizer

2011-08-19T07:11:00.000-07:00

To listen to old vinyl records with a full sound, you need a diagram, called the RIAA equalizer. This can be found in older amps, but in the modern household equipment it no longer inserted. If you want to archive your vinyl records on your PC, you will also need RIAA equalizer. It would be nice if the corrector was built in power amp for small speakers or headphones. In the device described here is an amplifier. It consists of two parts: a proof-reader and an amplifier.

Corrector is made for ultra-low noise IC NE5532 . In the scheme of the corrector uses metal-film resistors with a tolerance of 1%, power 0.6 W, the capacitors must be within a tolerance of 5% or better, with an operating voltage 63 ... 100 V. The corrector has direct access to an external amplifier or PC.

The amplifier chip is made on the LM1877 . It provides a power output of 2 watts per channel with very low distortion. Potentiometer P1 is used to adjust the output power amplifier.

The whole scheme is powered by an external DC voltage 12 ... 16 V. The scheme, photos of the device and the PCB can be downloaded on the appropriate links.

Parts list :

Component	Type			Number

Resistor	1%	27 Ohm	0.6 W	2
Resistor	1%	1.1 kOhm	0.6 W	2
Resistor	1%	10K	0.6 W	2
Resistor	1%	22 kOhm	0.6 W	2
Resistor	1%	33k	0.6 W	1
Resistor	1%	47K	0.6 W	2
Resistor	1%	100K	0.6 W	4
Resistor	1%	1 M	0.6 W	6
Potentiometer	2 × 50 kOhm logarithmic			1
Capacitor	5%	1.5 nF	63 V	4
Capacitor	5%	3.3 nF	63 V	2
Capacitor	10%	100 nF	63 V	4
Capacitor	10%	220 nF	63 V	2
Capacitor	10%	470 nF	63 V	2
Electrolytic capacitor	5 × 11 mm	1 uF	50	2
Electrolytic capacitor	4 × 7 mm	10 uF	35	4
Electrolytic capacitor	5 × 11 mm	33 uF	35	2
Electrolytic capacitor	6 × 11 mm	100 uF	25 V	1
Electrolytic capacitor	10 × 20 mm	1000 uF	25 V	1
Electrolytic capacitor	10 × 30 mm	2200 uF	16	2
IC	LM1877N-9			1
IC	NE5532			1
Zener diode	BZX55C9V1	9.1	0.5 W	1
RCA connector for mounting on the H / V, single, red (right channel)				2
RCA connector for mounting on the H / V, single, white (left channel)				2
The power connector for mounting on the p / p 5 × 2.5 mm				1
Headphone jack for mounting on the p / p				1
External power supply 12 V / 5 W minimum				1

5 band graphic equalizer using single BA3812L IC

2009-03-24T17:01:00.000-07:00

5 band graphic equalizer

This circuit uses an individual chip, IC BA3812L to realize the five band graphic equalizer for use in HI-FI audio systems. The BA3812L five-point graphic equalizer that has all the necessary functions integrated into one IC. This IC
consists of five tone control circuit and input and output buffer amplifier. The BA3812L features low distortion, low noise and wide dynamic range, and is an ideal choice for Hi-Fi stereo applications. It also has a wide operating voltage range (3.5V to 16V), which means that it can be adapted for use with most types of stereo equipment. The five center frequencies are independently set using external capacitors, and as an output stage buffer amplifier and tone control circuits independent clause, fine control over part of the frequency band is possible, with two BA3812Ls, it is possible to build a 10-point graphic equalizer. The number increases and the cut can be set by external components. The recommended power supply is 8V, but the circuit should work for 9V supply as well. Voltage limit is 16V. Circuit diagram is given in operation for around five frequency bands:

100Hz
300Hz
1kHz
3kHz

5 Band Graphic Equalizer

2009-03-24T16:32:00.000-07:00

Other units have a graphic equalizer. five bands. is fundamentally different from other circuits that use instead of IC, the transistor and the power go up / - 24V DC, for low distortion. With the switch S1, we can isolate the EQ. inside or outside the service, so that the signal is not a passport is not altered.

PART LIST
R1-4-37 =	330 Kohms	C4=	220uF 63V
R2-7=	2.2Kohms	C5=	1uF 100V
R3=	120Kohms	C6=	220nF 100V
R5-6=	6.8Kohms	C7=	47nF 100V
R8....12=	1.8Kohms	C8=	15nF 100V
R13....15=	200Kohms	C9=	5.6nF 100V
R16 =	180Kohms	C10=	47nF 100V
R17 =	150Kohms	C11=	15nF 100V
R18 =	1.5Kohms	C12=	4.7nF 100V
R19....22=	1Kohms	C13=	1.5nF 100V
R23....27=	220Kohms	C14=	470pF 100V
R28....32=	15Kohms	C15=	68pF 100V
R33 =	10Kohms	C16-19=	100nF 100V
R34 =	15Mohms	C17-20=	470uF 40V
R35-38 =	220 ohms 1W	C18=	4.7uF 63V
R36 =	100 ohms	RV1....5 =	50Kohms Lin.
C1-3 =	1uF 40V	Q1.....7 =	BC550C
C2=	68pF 100V	S1=	2X2 SW

High-End Audio Modular Preamplifier

2009-03-24T10:57:00.000-07:00

Generally

This preamplifier is requirement result of many friends to give a high quality preamplifier, capable to drive high quality power amplifiers with good sound, it's not however difficult to making, it combines simplicity and handiness. It does not allocate a lot of facilities and various filters, but is drawn so the acoustic signal follows the smaller course without it's influenced by stages they do not need, with result the coloration of reproduced sound. It follows the modular philosophy that gives the possibility of changing each module and adapted in yours acoustic taste. Thus it will constitute the base for long-standing listening and experimentations. Use relay in the input selection ensures the smaller way and the better possible quality, because the relay contacts are not influenced easily by temperature changes and environment, offer smaller ohmage and parasitic resistance in the signal course. This of course depends also from the relay quality that you will select; I propose choice of relay small size and very high quality. Enough companies they manufacture such relays. Exists different possibility of choice for the Listen inputs and different for the analogue recording. Thus we can listen other sound source and other recording. Exists the possibility, for those who they judge of suppressing Balance potesometer [RV1], the recording selector and proportional relays making the manufacture simplest and more economic. In the each place module we have big variety of choices. Certain you I will give I below, but can make also your choices. As you will observe exist somebody "facilities" as monophonic [MONO] operation etc.
Input selector

Main Preamplifier

Discription

The choice of acoustic signal follows two streets [Fig.1A]. First with the selector S1 that is 1 contact 12 places, from which I use five. Exists the possibility to using also the remainder places adding to them proportional relay and Led. Each place supplies the proportional relay and Led that it shows whom input has been selected. The supply becomes with + 12V [+V4]. Selector S2 functions with the same way as the S1, supplying them proportional relays and Led, so that becomes the source choice that recording. Afterwards recording relay exists the buffer unit round IC1 [Fig.1B]. In the part list exist certain choices for opamp. that will undertake this role, but exists the possibility of selecting somebody of your choice. All the inputs are high level apart from input PHONO [RIAA Filter] that is directed in proportional module PHONO [J11] and by his exit return again in the block selector relay. He is alone module that has stereophonic structure and the two channels that abut to connector [J11]. Those who do not allocate the old turnable [pick-up] and disks collection of 33 turns, they will not need PHONO module. It can easily they change him in line, suppressing connector J11 and you connect the corresponding inputs J1-2 in proportional contacts P1-2 relay RL1-6. The relay team of listening direct the stereo acoustic signal from some input in Buffer module [J12-13]. The gain of this unit is X1, does not provide no gain in the signal. They follow potesometer BALANCE [RV1a-b] and VOLUME [RV2a-b] that regulates the balance and the level of acoustic signal respectively. These two Potentiometers should be high quality also from acquaintance constructor. Afterwards the Potentiometer VOLUME exist the MUTE stage that is constituted by a divider R21-22 and the contacts of relay RL11 that are found parallel with the R21. In normal operation the contact jumper the R21, the signal pass without fall. If it close switches S3, turns on the D22, the RL11 arms, the contact opens and the R21 with the R23 they configure an divider which create a fall of level in acoustic signal [-20dB]. The same logic is also for the other channel where exist the resistors R23-24. In the continuity the acoustic signal enter via the J16-17 in Line module what offers the suitable gain in the signal and this drive all the goods power amplifiers. Then exist the contacts of RL12 that the operation of is described in the delay circuit.
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Low Noise Microphone Preamplifier

2009-03-19T16:10:00.001-07:00

This is a design for a low noise microphone preamplifier, which is ideally suited to low impedance (600 Ohm nominal) microphones. One limitation is that it is not balanced, which is not a problem in a home recording environment, but will allow the mic lead (and case) to pick up noise with long cable runs or in a hostile environment.http://sound.westhost.com/project13.htm

Stereo Tube Amplifier

2009-03-19T13:18:00.000-07:00

By Weslee Kinsler

The circuit is simple, yet is capable of excellent performance. I designed it specifically for use as an amplifier for the digital sound card in my computer. Audio input can be from any two-channel line level device such as a television, CD player, or VCR. It is of the tube type, using only 5 tubes total with no more than about 45 Watts power consumption from the outlet. It uses 3 types of tube 1 5Y3 GT vacuum rectifier, 2 6SF5 GT high-mu triodes, and 2 6K6 power beam amplifiers. These are all full-size octal type tubes which are commonly available today for between $3-5 each.

Schematic

Parts [Qty]

R1, R10, R13[6]...........470K 1/2W
R2[2]...........1 Meg 1/2W
R4[2]...........220K 1/2W
R5[2]...........330 Ohm 2W
R6[2]...........220K 1/2W
R7[2]...........2.2Meg 1/2W

R8[2] ..........1Meg 1/2W
R9[1]...........720 Ohm 20W
R11[2]..........33K 1/2W
R12[2] .........22K 1/2W
C1, C9[4] .....0.005uF 400V Capacitor
C2[2]...........0.05uF 600V
C3[2]...........20uF 25V Electrolytic
C4 [2]...........0.01uF 400V Capacitor
C5[2]...........200uuF 400V Ceramic Disc
C6, C7[2].....15uF 450V
C8[2] .........15uF 400V
T1[1]..........117V Primary, 350VCT Secondary, 6.3V Secondary, 6.3V Secondary
T2[2]..........7600 Ohm Primary, 4 or 8 Ohm Secondary
SW1[1]..........SPST Switch
SP1, SP2[2]..........12" or smaller, 4 or 8 ohm speakers
MISC 1 [5]...........tube sockets, 2 RCA jacks, PC board or chassis, wire, knobs, etc.

Notes:

1. Email Weslee Kinsler with questions, comments, etc.
2. The 6V6 GT tube may be substituted for the 6K6 to lower power requirements.
3. C8 is for radio interference suppression and may be omitted.
4. The 5Y3 GT tube should be mounted in a vertical position and be well ventilated. The 6K6 and 6SF5 tubes can be mounted in any position.
5. Controls should have an audio taper.

source: http:/www.aaroncake.net/

Preamplifier for 60W Bass Amplifier

2009-03-19T11:50:00.000-07:00

Part
P1_____________10K Linear Potentiometer
P2_____________10K Log. Potentiometer

R1,R2___________68K 1/4W Resistors
R3_____________680K 1/4W Resistor
R4_____________220K 1/4W Resistor
R5_____________33K 1/4W Resistor
R6_____________2K2 1/4W Resistor
R7_____________5K6 1/4W Resistor
R8,R18__________330R 1/4W Resistors
R9_____________47K 1/4W Resistor
R10____________18K 1/4W Resistor
R11____________4K7 1/4W Resistor
R12____________1K 1/4W Resistor
R13____________1K5 1/4W Resistor
R14,R15,R16______100K 1/4W Resistors
R17____________10K 1/4W Resistor

C1,C4,C8,C9,C10__10µF 63V Electrolytic Capacitors
C2____________47µF 63V Electrolytic Capacitor
C3____________47pF 63V Ceramic Capacitor
C5____________220nF 63V Polyester Capacitor
C6____________470nF 63V Polyester Capacitor
C7____________100nF 63V Polyester Capacitor
C11___________220µF 63V Electrolytic Capacitor

Q1,Q3_________BC546 65V 100mA NPN Transistors
Q2___________BC556 65V 100mA PNP Transistor
J1,J2_________6.3mm. Mono Jack sockets
SW1__________SPST Switch

Class-A/AB Amplifier

2009-03-19T11:37:00.000-07:00

Notes:
This circuit developed out of my 30+ years of JLH class-A based investigations.

The original 'simple' 1969 JLH class-A design provided excellent first cycle accuracy through mid and high frequencies (dynamic clarity) because there were no stabilisation components nor a series output choke, whilst the NFB error correction was established via the input emitter (some describe this as current feedback).

However it did generate rather a lot of heat, the damping was 'soft' at lower frequencies and the positive slew was weaker in the presence of loudspeaker system generated back-EMF at higher audio frequencies. I was eventually able to improve upon the original JLH circuit by;-

1) adding a 'helper' class-AB output stage to substantially increase power and efficiency, but leave sufficient class-A bias for real-time control maintenance through phase shifted zero current class-AB crossovers;

2) adding a differential input stage for zero output voltage control, but also fit a 10nF base-emitter capacitor for differential voltage operation at audio frequencies, though with leading emitter routed feedback to maintain circuit stability and NFB control above 20kHz;

3) adding a current mirror to obviate power-up thump, but also fit a 220nF base-emitter capacitor to ensure stable 'source only' operation above audio frequencies.

Thus, when compared to conventional circuits, one complete high frequency phase change has been removed from the closed feedback loop in order to minimise need for any additional stabilisation components that would otherwise render the circuit inductive at audio frequencies. The original JLH hf (current feedback) stability and simultaneously phase coherent class-A control are retained.

This circuit was intended for 2SC5200 operation, yet it has been successfully constructed using other device types, including a single Sanken 2SA1216+2SC2922 pair in place of the paralleled ABs. Keep VAS, Zobel and output related wires 5cm/2" away from input devices and wiring. Use a star earth from the input location. Use star power distribution from each fused 10mF at the pcb. Use star output node connections. Parallel all large Cs with smaller ones. Do not twist any extender e-b-c wires that might be used out to heatsink mounted output stage devices. Mount the Vbe multiplier on the output heatsink for automatic temperature compensation. Adjust the class-AB pre-set to 50%, but adjust the class-A pre-set to be a short circuit before the initial switch-on. I recommend 22 ohm per rail in place of the fuses in case of error at initial power-up, but do not try to set any bias with them in circuit.

100W-4ohm (conservatively rated) is 50W-8ohm. For 70W/35W use 30V rails. For 50W/25W, 25V. For 100W-8ohm with higher voltage rails try a 47pF Miller connected C.dom to the VAS, it should compensate for device/stability changes with increased voltage, yet not adversely affect the class-A operation in maintaining crossover continuity. Always bias from zero current if you increase the rail voltage. Insert a 0.22 ohm series output resistor to maintain stability when driving purely capacitive loads, and don't forget to bi- or tri-wire out to composite loudspeaker system sections and drivers in order to obviate single cable developed dynamic voltage drops with varying loudspeaker system generated back-EMFs, no matter how expensive your loudspeaker cable might be.

Graham Maynard.

Source:www.zen22142.zen.co.uk1999-2009. cxem.net. A

60W Bass Amplifier

2009-03-19T11:06:00.000-07:00

Low-cut and Bass controls
Output power: 40W on 8 Ohm and 60W on 4 Ohm load

Amplifier parts:
R1______________6K8 1W Resistor
R2,R4___________70R 1/4W Resistors
R3______________2K 1/2W Trimmer Cermet
R5,R6___________4K7 1/2W Resistors
R7______________220R 1/2W Resistor
R8______________2K2 1/2W Resistor
R9______________50K 1/2W Trimmer Cermet
R10_____________68K 1/4W Resistor
R11,R12_________R47 4W Wirewound Resistors

C1,C2,C4,C5______47µF 63V Electrolytic Capacitors
C3______________100µF 25V Electrolytic Capacitor
C6______________33pF 63V Ceramic Capacitor
C7______________1000µF 50V Electrolytic Capacitor
C8______________2200µF 63V Electrolytic Capacitor (See Notes)

D1______________LED Any type and color
D2______________Diode bridge 200V 6A

Q1,Q2___________BD139 80V 1.5A NPN Transistors
Q3______________MJ11016 120V 30A NPN Darlington Transistor (See Notes)
Q4______________MJ11015 120V 30A PNP Darlington Transistor (See Notes)

SW1____________SPST Mains switch
F1______________4A Fuse with socket
T1______________220V Primary, 48-50V Secondary 75 to 150VA Mains transformer
PL1____________Male Mains plug
SPKR___________One or more speakers wired in series or in parallel
Total resulting impedance: 8 or 4 Ohm
Minimum power handling: 75W

25 Watt MosFet Audio Amplifier

2009-03-19T11:01:00.000-07:00

High Quality simple unit No need for a preamplifier
Parts: R1,R4________47K 1/4W
Resistors R2________4K7 1/4W
Resistor R3_________1K5 1/4W
Resistor R5________390R 1/4W
Resistor R6________470R 1/4W
Resistor R7_________33K 1/4W
Resistor R8________150K 1/4W
Resistor R9_________15K 1/4W
Resistor R10________27R 1/4W
Resistor R11_________500R 1/2W
Trimmer Cermet R12,R13,R16__10R 1/4W
Resistors R14,R15___220R 1/4W
Resistors R17_________8R2 2W
Resistor R18__________R22 4W
Resistor (wirewound) C1__________470nF 63V
Polyester Capacitor C2____________30pF 63V
Polystyrene Capacitor C3,C5______470µF 63V
Electrolytic Capacitors C4,C6,C8,C11_100nF 63V
Polyester Capacitors C7__________100µF 25V
Electrolytic Capacitor C9__________10pF 63V
Polystyrene Capacitor C10_________1µF 63V
Polyester Capacitor Q1-Q5______BC560C 45V 100mA
Low noise High gain PNP Transistors Q6_________BD140 80V 1.5A PNP
Transistor Q7_________BD139 80V 1.5A NPN
Transistor Q8_________IRF532 100V 12A N-Channel Hexfet
Transistor Q9_________IRF9532 100V 10A P-Channel Hexfet Transistor

50 Watt Amplifier

2009-03-19T10:55:00.000-07:00

This is a handy, easy to build general purpose 50 watt amp. The amp has an input for a radio, TV, stereo or other line level device. It also has a phono input for a record player, guitar, microphone or other un-amplified source. With the addition of a low pass filter at the input, it makes a great amp for a small subwoofer.

Part :
R1 ....................... 200 Ohm 1/4 W Resistor
R2 ....................... 200K 1/4 W Resistor
R3 ....................... 30K 1/4 W Resistor
R5 ....................... 1K 1/4 W Resistor
R6 ....................... 5K 1/4 W Resistor
R7,R10 ............... 1 Meg (5%) 1/2 W Resistor
R8,R9 ................. 0.4 Ohm 5 W Resistor
R11 ..................... 10K Pot
R12,R13 ............. 51K 1/4 W Resistor
R14 ..................... 47K 1/4 W Resistor
C1 ....................... 100uF 35V Electrolytic Capacitor
C2 ....................... 0.011uF Capacitor
C3 ....................... 3750pF Capacitor
C4,C6 ................. 1000pF Capacitor
C5,C7,C8 ........... 0.001uF Capacitor
C9 ....................... 50pF Capacitor
C10 ..................... 0.3uF Capacitor
C11,C12 ............. 10,000uF 50V Electrolytic Capacitor
U1,U2 ................. 741 Op Amp
U3 ....................... ICL8063 Audio Amp Transister Driver thingy
Q1 ....................... 2N3055 NPN Power Transistor
Q2 ....................... 2N3791 PNP Power Transistor
BR1 .................... 250 V 6 Amp Bridge Rectifier
T1 ....................... 50V Center Tapped 5 Amp Transformer
S1 ....................... SPST 3 Amp Switch
S2 ....................... DPDT Switch
F1 ....................... 2 Amp Fuse
SPKR1 ................ 8 Ohm 50W Speaker
MISC ................... Case, Knobs, Line Cord, Binding Posts Or Phono Plugs (For Input And Output), Heatsinks For Q1 And Q2