Speech Eroder

Nowadays, the speech
quality on our telephone systems is generally very good, irrespective of
distance. However, there are occasions, for instance, in an amateur
stage production, or just for fun, when it is desired to reproduce the
speech quality of yesteryear. The eroder circuit accepts an acoustic
(via an electret micro-phone) or electrical signal. The signals are
applied to the circuit inputs via C1 and C2, which block any direct
voltage. The input cables should be screened. The signals are brought to
(about) the same level by variable potential dividers P1-R1-R4 and
P2-R2-R3, and then applied to the base of transistor T1. The level of
the combined signals is raised by this preamplifier. The preamplifier is
followed by an active low-pass filter consisting of T2–T4, C3, C4,
红包扫雷苹果下载地址 R6–R8, and P4.

Circuit diagram:

Speech Eroder Circuit

Speech Eroder Circuit Diagram

Although, strictly speaking, P3 serves merely to adjust the volume
of the signal, its setting does affect the filter characteristic. Note,
by the way, that the filter is a rarely encountered current-driven one
in which C3 and C4 are the frequency-determining elements. It has a
certain similarity with a Wien bridge. Transistors T3 and T4, and
resistors R8 and P4 form a variable current sink. The position of P4
determines the slope of the filter characteristic and the degree of
overshoot at the cut-off frequency. The low-pass filter is followed by
an integrated amplifier, IC1, whose amplification is matched to the
input of the electronic circuits connected to the eroder with P5. The
final passive, third-order high-pass filter is designed to remove
frequencies above about 300 Hz. The resulting output is of a typical
红包扫雷苹果下载地址 nasal character, just as in telephones of the past.

Author: T. Giesberts
红包扫雷苹果下载地址 Copyright: Elektor Electronics

Voice Scrambler

With this circuit you
can modify how your voice sounds by changing the pitch of your voice.
This circuit can be connected to a phone and with a duplicate circuit on
the end of the phone line, you can have a scrambled voice
communication. The way the circuit works is as follows: If we cut the
circuit in half at the T2 transformer and include the LM324 on the left
side, you will see that the LM324 portion of the circuit is a tone
oscillator which shifts the frequency of all input signals to a new
higher frequency. When the voice and the tone oscillator mix frequencies
the voice is not recognized. The voice signal is then inputted to the
second stage which again shifts the voice signal again. I recommend that
the first stage be tuned to a frequency that is 100hz lower then the
红包扫雷苹果下载地址 second stage.

Tracking Down Scratchy Pots

One of the most common
faults in audio equipment is noisy pots – potentiometers that introduce
scratching or crackling noises into the signal as they are adjusted. The
problem is that sometimes a perfectly good pot will sound scratchy or
crackly because of an intermittent connection or because DC is getting
into it through a faulty capacitor or an out of balance direct-coupled
stage. So how can you determine whether a pot really is scratchy before
going to the trouble of finding and fitting a physically compatible
红包扫雷苹果下载地址 replacement?

This solution is simple and involves a test setup which can be done
with the pot still in circuit (but with the power off). Using clip leads
or temporarily soldered wires connected directly to the pot’s
terminals, connect the pot as a volume control between a signal
generator and a signal tracer (or audio amplifier), as shown. Then
adjust the pot up and down. If the signal tracer gives scratchy noises
on top of the tone from the signal generator, then the pot is faulty.

author: andrew partridge – copyright: silicon chip electronics

Symmetric Noise Source

If a transistor junction
operating in Zener breakdown is used as a noise source, the amplitude
of the noise signal is asymmetric. This problem can be solved by using
two transistors as two independent noise sources. One of these has a
series resistor to earth, and the other has a series resistor to the
supply line. Each of these noise sources produces an asymmetric noise
voltage, with opposite asymmetry. If these two voltages are combined,
the amplitude of the result will be symmetric. In the circuit diagram,
T1 and T2 are the noise sources. The series resistors are R2 (to earth)
and R4 (to the positive supply line).

The supply voltage for the noise sources has been made adjustable,
to allow the noise generation of the transistors to be optimized. This
is because the amount of noise produced depends on the power supply
voltage. P1 and R1 provide an adjustable supply voltage between 8 and 12
V for the noise stage around T1, while P3 and R3 perform the same
function for T2. C3 and C5 smooth these voltages. Since the amplitudes
of the two noise sources will never be the same, it is necessary to take
a weighted sum of the two signals. Consequently, P2 is included between
the outputs of the noise sources as a sort of balance control.

Since the DC levels of the two noise sources are not the same, C4 is
also included in the balance network. The weighted sum of the two
signals is present on the wiper of P2, superimposed on the DC signal of
noise source T1. This DC level is also used for the DC bias of the
buffer stage T3. The buffer isolates the noise sources from whatever
circuit is connected to the output. To adjust the circuit, connect an
oscilloscope to the output. First, turn P2 all the way to the left. Now
rotate P1 until a maximum noise signal is seen on the oscilloscope.
Next, turn P2 all the way to the right, and then adjust P3 for the best
noise signal. Finally, adjust P3 so that the noise signal looks
symmetric. The circuit provides an output voltage of approximately 150mV
pp. The current consumption is 2mA. The oscilloscope shows the
asymmetric noise signal on channel 2, and the symmetric noise signal on
红包扫雷苹果下载地址 channel 1.

Wireless Digital Audio Streaming IC

Wireless Digital Audio Streaming IC

Wireless Digital Audio Streaming IC

红包扫雷苹果下载地址2.4 ghz audio ics that transmit uncompressed cd-quality wireless audio over a rock solid rf (radio frequency) link, with error-free, lossless transmission. the solutions target consumer, portable and high-end audio applications, such as wireless headphones, headsets and speakers.

CC85xx key benefits:
  • Uncompressed CD-quality wireless audio
  • Rock solid 16-bit 44.1/48kHz RF link
  • Multichannel and multipoint capabilities: streaming of up to four simultaneous channels
  • USB support
  • Best-in-class coexistence with Bluetooth®, WLAN and other 2.4 GHz devices
  • Easy-to-use designer’s PC software tool
  • Works out of the box with selected TI audio devices

the cc8520 (two-channel) and cc8530 (four-channel) system-on-chips (soc) are complete integrated solutions with rf protocol, microcontroller, audio codec setup support, application designs and free purepath configurator pc software tool. additionally, the cc8521 and cc8531 socs contain usb audio support for all major operating systems.

ti’s purepath wireless headset reference design is the market’s most cost-effective design for high quality headsets and headphones. the design has a low electronic bill-of-material cost and achieves a 22-hour life on a 465 mah battery – a 100 percent increase as compared to currently available headsets.

the purepath wireless cc8521 and cc8531 usb dongle reference design allow developers to design low-cost, small usb dongles for wireless audio applications, without code development or in-depth understanding of usb protocols.

Digital Input 2 W Class-D Audio Amp

Digital Input 2 W Class-D Audio Amp

Digital Input 2 W Class D Audio Amp

红包扫雷苹果下载地址the ssm2518 is a digital input, class-d power amplifier that com-bines a digital-to-analog converter (dac) and a sigma-delta (σ-δ) class-d modulator. this unique architecture enables extremely low real-world power consumption from digital audio sources with excellent audio performance. the ssm2518 is ideal for power sensitive applications, such as mobile phones and portable media players, where system noise can corrupt small analog signals such as those sent to an analog input audio amplifier.

using the ssm2518, audio data can be transmitted to the amplifier over a standard digital audio serial interface, thereby significantly reducing the effect of noise sources such as gsm interference or other digital signals on the transmitted audio. the closed-loop digital input design retains the benefits of an all digital amplifier, yet enables very good psrr and audio performance. the three level, σ-δ class-d modulator is designed to provide the least amount of emi interference, the lowest quiescent power dissi-pation, and the highest audio efficiency without sacrificing audio quality.

红包扫雷苹果下载地址input is provided via a serial audio interface, programmable to accept all common audio formats including i2s and tdm. control of the ic is provided via an i2c control interface. the ssm2518 can accept a variety of input mclk frequencies and can use bclk as the clock source in some configurations.

additional features include a soft digital volume control, de-emphasis, and a programmable digital dynamic range compressor.

the architecture of the ssm2518 provides a solution that offers lower power and higher performance than existing dac plus class-d solutions. its digital interface also offers a better system solution for other products whose sole audio source is digital, such as wireless speakers, laptop pcs, portable digital televisions, and navigation systems.

APPLICATIONS
  • Mobile phones
  • Portable media players
  • Laptop PCs
  • Wireless speakers
  • Portable gaming
  • Small LCD televisions
  • Navigation systems

Digital Audio Platform

Digital Audio Platform

Digital Audio Platform

atmel® corporation, a leader in microcontroller and touch technology solutions, announced a complete digital audio platform for consumer, automotive and industrial applications. the atmel digital audio platform offers audio equipment and mobile accessory oems (original equipment manufacturers) a complete hardware and firmware solution that greatly simplifies the task of designing high-quality digital audio equipment. the new digital audio platform is implemented with the atmel avr® uc3 microcontrollers specifically tailored for audio applications such as smartphone and media player docking stations.

the digital audio platform integrates dedicated microcontrollers, evaluation kits and firmware ip. the firmware ip includes control and streaming interfaces for a selection of popular smartphones and portable media players as well as mp3, wma and aac decoders to allow designers to decode compressed music files; usb protocol stacks; and a complete file system to allow designers to navigate through mass storage devices such as usb flash disks or sd cards. the platform uses a ready-to-go commercial licensing model for firmware ip that enables designers to keep their firmware code confidential, unlike many audio license agreements today that require the firmware code to be open. the digital audio platform is ideal for applications including docking stations, usb mass storage, sd card playback, car stereos, usb speakers, microphones, and various voice and music equipment. other applications include mobile accessories (accessories connected to smartphones or tablets) that use the same underlying technology.

with all the benefits of the atmel avr uc3 microcontrollers, the digital audio platform offers designers a system that is highest in performance and lowest in power. the avr uc3 microcontrollers also support the atmel qtouch® library for easy implementation of elegant, differentiating user interfaces for capacitive touch buttons, sliders and wheels.

“designing digital audio applications has historically been a complex, cumbersome task of finding the required firmware ip, licensing it and finally stitching it together in a working application,” said ingar fredriksen, sr. director of avr products, atmel corporation. “the atmel digital audio platform offers an ‘out-of-the-box solution’ to enable designers to develop today’s audio applications with minimum engineering effort spent on getting the core design working.”

to help accelerate a designer’s audio application development, evaluation kits (evk) are also available with the digital audio platform. evaluation kits include a board with an atmel avr mcu and free downloadable source code. the atmel avr evk1104au/1105au audio development kits are available in the atmel store at http://store.atmel.com and through atmel-qualified distributors.

HiFi Audio DSP

HiFi Audio DSP

HiFi Audio DSP

tensilica, inc. announced the immediate availability of dolby volume on its popular hifi audio dsps (digital signal processors). the implementation is based on software developed by dolby and has passed dolby’s certification. dolby volume enables designers of socs (systems on chips) for home entertainment systems and digital televisions, and for handsets with mobile dtv, to provide viewers with a consistent playback volume level across all sources and content.

红包扫雷苹果下载地址“viewers have been asking for a high-quality solution to the constant volume fluctuations encountered in commercial content, as they find themselves constantly reaching for the volume control to compensate. not only do these constant changes distract from the viewing experience, they also bring the potential for hearing damage when using earbud headphones,” stated larry przywara, tensilica’s senior director of multimedia marketing. “the dolby volume solution, optimized for our hifi dsps, ensures a reference-quality listening experience at any volume level.”

红包扫雷苹果下载地址“dolby volume on tensilica hifi audio dsps is an important addition to the wide range of dolby solutions available on tensilica products,” stated giles baker, senior director, broadcast, dolby laboratories. “as the number of designs based on tensilica’s hifi dsp increases, we are looking forward to even greater adoption of dolby volume.”

the technologies behind dolby volume, such as loudness-domain signal processing and auditory scene analysis, use advanced models of human hearing based on psychoacoustics, the science of human hearing. this makes it possible to correct source and content level differences without undesirable artifacts, such as pumping and breathing, introduced by previous compression and expansion techniques.

dolby volume also actively balances low, middle and high frequencies in each channel to compensate for the ear’s changing sensitivity as the playback level is raised or lowered. as a result, the nuances and impact of the original mix are maintained even at low playback levels.

tensilica’s hifi audio dsp provides chip designers and system oems with proprietary audio processing algorithms with one hardware platform that efficiently runs the most audio standards, as well as proprietary audio enhancement algorithms. tensilica, its customers and its partners have ported over 80 software packages to the hifi architecture, so designers can pick the software they need for the application. as the market evolves and new standards are defined, they can easily and quickly be ported to the hifi architecture, thereby “future proofing” the chip design.

“tensilica’s hifi architecture is designed to be extremely efficient for applications like dolby volume,” stated michael vulikh, ceo of p-product. “hifi’s c-based programming model enabled us to quickly develop an optimized port and meet tensilica’s aggressive delivery schedule.” tensilica’s audio partner, p-product, did the dolby volume port.

Digital Audio Processors

Digital Audio Processors

Digital Audio Processors

红包扫雷苹果下载地址stmicroelectronics, a global semiconductor leader serving customers across the spectrum of electronics applications and a leading supplier of audio ics, has introduced a digital audio processor for multi-microphone applications. st’s new smart voice product family combines superior processing performance with unparalleled scalability and programmability to bring dramatic sound-quality improvements to mobile phones, tablets, gaming devices and video security systems.

the smart voice audio processors unleash the powerful capabilities of digital mems microphones in multi-microphone arrays, especially for st’s mems microphones, enabling features like acoustic echo cancellation, noise suppression, or beam-forming, which are invaluable with the increasing use of cell phones and other consumer devices in noisy and uncontrollable environments. multi-microphone systems also address next-generation audio applications, such as 3d sound processing, sound-source localization, virtual microphones, and audio zooming.

the combination of st’s smart voice processors and digital mems microphones creates a powerful and flexible acoustic sub-system that brings crucial advantages to equipment manufacturers and users alike. manufacturers will benefit from a simplified system design, which reduces costs and time to market and facilitates end-product differentiation, while consumers will enjoy clearer conversations and longer battery life on their devices.

a technology breakthrough, st’s newest digital audio processor is based on an optimized hard-wired acoustic processing engine that offloads intensive computing tasks from the main application processor. it handles up to 6 digital microphone inputs with dynamic array re-configuration and dedicated channel processing. the device integrates a high-quality scalable acoustic processing core with a tunable 10-band equalizer, a peak limiter, and gain and volume controls. the data are output through a digital (i2s) or analog (pwm) interface.

st’s smart voice processors come in a bundle with ap workbench, a user-friendly programming tool, and together with the company’s mems microphones and sound terminal class-d audio amplifiers they form a complete one-stop offering for advanced sound-input applications.

the sta321mpl, the first device in st’s smart voice family, is sampling now. mass production is scheduled for q2 2012, with unit pricing at $4 for volumes in the range of 1,000 pieces. if your company has a high-volume need, please contact your st sales office.

2002 Ford Escort Fuse Box Diagram

2002 Ford Escort Fuse Box

2002 Ford Escort Fuse Box Diagram

2002 Ford Escort Fuse Box Map

2002 Ford Escort Fuse Box Map

Fuse Panel Layout Diagram Parts: 红包扫雷苹果下载地址meter asc, mirror, rear wiper, door lock, stop lamp, hazard warning, horn, tail light relay, room, air conditioner, wiper, cigar lighter, fog lamp, fuel injection, air bag, audio, power window, heater, engine.