Dsp

3D IMAGE GENERATION USING ADSP

The DSP processors are commonly used to handle the image

data, where the image has to be processed digitally and

accurately, such as the application like scanning systems in Bio-

medial Field.

The processor has powerful instruction sets and most of the

instructions are one cycle execution. So that the result can be

obtained much faster and efficient, which is required for Analog

signal processing.

In this project, we are generating a 3D image using ADSP (DSP)

processor. The generated Image can be rotated in 3-axis (X,Y &

Z) using flexibility of the assembly instruction set of the processor.

Initially we draw the Image in a graph sheet and take X,Y co-

ordinates of the image. This data will be input to our systems

This can be obtained from the sensor used for scanning in Real

time

The Software is developed such that it can take these data and

generates the X & Y deflection signal.

The hardware has a DSP with Boot memory. The developed

firmware will be loaded in EPROM and it will be configured as

Boot memory for DSP.

On reset the Processor loads the data (program) from the Boot

memory to its internal Program memory and starts executing the

Program.

Two DACs (Digital to Analog Converter) are used to control the X

& Y deflection of CRT. So the software controls these two DACs

and generates the Image in CRT.

We can also use a CRO as CRT in X-Y mode. The signal given to

the Channel1 of CRO may deflect the dot X-direction and

Channel2 for Y-direction.

ADSP

Adaptive Digital Signal Processing; exclusive to SGC, a DSP

technology which processes the signal to eliminate unwanted

noise and improve incoming signal

DSP

DSP stands for Digital Signal Processor. DSP chips are widely

used in sound cards, fax machines, modems, cellular phones,

high-capacity hard disks and digital TVs

DSP chips are used in sound cards for recording and playback

and speech synthesis. Other audio uses are amplifiers that

simulate concert halls and surround-sound effects for music and

home theater

Digital Signal Processing uses mathmatics to operate on a digital

signal (such as a digital audio stream) to generate some type of

altered output. DSP is used heavily in software and hardware

effects processing.

DSP chips are found on an increasing number of sound cards to

provide extra audio processing power and help relieve the

computers CPU of this type of work, much like a 3D graphics

accelerator would for rendering 3D graphics

3D IMAGE GENERATION USING ADSP

BLOCK DIAGRAM

BOOT

MEMORY

ADSP

PROCESSOR

DAC 1

8Bit

DAC 2

8Bit

TO CRO

CH1

TO CRO

CH2 DATA

MEMORY

ADAPTIVE FILTER IMPLEMENTATIONADAPTIVE FILTER IMPLEMENTATIONADAPTIVE FILTER IMPLEMENTATIONADAPTIVE FILTER IMPLEMENTATION

The aim of the project is to develop a digital system to perform adaptive

filtering of analog signals.

Adaptive filters are used to accomplish a variety of applications,

including

1. Echo cancellation in Voice or Data signals.

2. Noise cancellation in Speech, Audio and Biomedical signal

processing.

3. Channel equalizations in Data communication to minimize the effect

of intersymbol interference

BASICS OF ADAPTIVE DIGITAL FILTERS

Adaptive filter can be implemented from the FIR filters. Normally, in FIR

filters, the filter coefficients are constant. But in the adaptive FIR filters,

the filter coefficients are adjusted to minimize the errors.

Thus when we employ adaptive filter for echo cancellation, the FIR filter

coefficients are adjusted in every cycle to remove the echo from the

input. Adaptive filters can be implemented using the least Mean-square

(LMS) algorithm.

HARDWARE

The block diagram of ADSP system used for adaptive filtering is shown

in Fig.1. The system consists of ADSP as CPU, CODEC, EPROM,

RAM, Amplifier circuits, MIC and speaker. The CODEC has been used

for analog input and output.

The CODEC has been interfaced to ADSP through its serial port. The

optional hardware includes PC interfacing circuit consisting of serial

Input port and RS 232 level converter. The additional serial Input port

have been developed using buffer.

FUNCTIONING

The analog input signal from MIC are amplified and fed to CODEC. The

CODEC converts the analog signal into digital data and transfer the

data to DSP microprocessor.

The processor adds echo to the input and it will output the signal with

echo to the speaker.

The signal with echo can be stored in memory and then adaptive filter

can be introduced so as to remove the echo from the input signal.

After introducing the adaptive filter, we can hear that the echo is slowly

removed.

The adaptive filter coefficients are also output to CODEC which

converts the digital filter coefficients to analog and they can be viewed

on the CRO.

APPLICATIONS OF ADAPTIVE FILTER

� Digital Audio

� Digital Graphical Equalizer

� Noise removal

� High Speed Modems

� Noise removal in ECG signals

ADSP


technology which processes the signal to eliminate unwanted noise

and improve incoming signal

DSP

DSP stands for Digital Signal Processor. DSP chips are widely used in

sound cards, fax machines, modems, cellular phones, high-capacity

hard disks and digital TVs

DSP chips are used in sound cards for recording and playback and

speech synthesis. Other audio uses are amplifiers that simulate

concert halls and surround-sound effects for music and home theater



altered output. DSP is used heavily in software and hardware effects

processing.


provide extra audio processing power and help relieve the computers

CPU of this type of work, much like a 3D graphics accelerator would

for rendering 3D graphics

PCRS 232c

LEVEL

CONVERTER

SYSTEM BUS

BUFFER DAC CRO

Optional

Hardware

AMP

SERIALINPUT

PORT

Power Amp

Pre Amp

Speaker

Mike

CODEC ADSP

-2105

EPROM RAM

Fig 1. Block diagram of ADSP -2105 system used for adaptive filte

ring

DTMF GENERATION AND

DETECTION

The aim of the project is to develop a digital system for generation and

detection of DTMF tones. The DTMF (Dual Tone Multifrequency) are used

in Modern push button telephones.

The push button telephones are replacing the conventional dial-pulse

signaling in telephone networks worldwide.

PRINCIPLE

The DTMF tones are sinusoid signals of standard frequencies defined by

CCITT (Consultative Committee for International Telephones and

Telegraphy).

In push-button telephones, for each digit, two tones are assigned. One

tone is chosen out of 4 Row tones and the other is chosen out of 4 column

tones.

By this combination, 16 different DTMF digits can be formed as

shown in figure below.

The DTMF tones can be easily generated or detected using DSP

microprocessor based system.

DTMF tones can be generated using function generation programs and

DTMF tones can be detected by performing FFT analysis of the signals.

The FFT of a signal will give the frequency components of the signal and

from the knowledge of frequency component, the DTMF digit can be

identified.

HARDWARE

BLOCK DIAGRAM OF ADSP

DTMF GENERATION AND DETECTION

The system consists of ADSP-2105 as CPU, CODEC, EPROM, RAM,

Amplifier circuits, MIC and speaker. The CODEC has been used for

analog input and output, the CODEC has been interfaced to ADSP-2105

through its serial port.

The optional hardware includes PC interfacing circuit consisting of serial

Input port and RS 232 level converter. The additional serial Input port

have been developed using buffer.

FUNCTIONING

1. Generation

The generation program can be permanently stored on EPROM. The

program can be executed using the INTERRUPT key and on pressing

the interrupt key each time, the tones of different frequencies are

generated one by one.

For generation of each tone, the processor determines the digital

signal using function approximation technique and output to CODEC.

The CODEC converts the digital signal to analog signal, which is

amplified, and output to the speaker.

2. Detection

For detection of DTMF tones, the DTMF detection program is

permanently stored in another EPROM. The tone to be detected is

input to the system through CODEC.

The CODEC digitizes the analog signal and send the digital signal to

ADSP-2105. The processor performs FFT operation on the signal to

determine the frequency spectrum of the signal.

The spectrum is output to CRO through DAC and viewed on the screen

of CRO. From the knowledge of frequency spectrum, the input DTMF

tone can be identified.

APPLICATIONS

DTMF is popular in interactive control applications like

1. Telephone banking

2. Electronic Mail system

3. Interactive Telephone Answering systems

4. Railway Reservation system.

ADSP

Adaptive Digital Signal Processing; exclusive to SGC, a DSP technology

which processes the signal to eliminate unwanted noise and improve

incoming signal

DSP

DSP stands for Digital Signal Processor. DSP chips are widely used in sound

cards, fax machines, modems, cellular phones, high-capacity hard disks and

digital TVs. DSP chips are used in sound cards for recording and playback and

speech synthesis. Other audio uses are amplifiers that simulate concert halls

and surround-sound effects for music and home theater

Digital Signal Processing uses mathmatics to operate on a digital signal

(such as a digital audio stream) to generate some type of altered output.

DSP is used heavily in software and hardware effects processing.

DSP chips are found on an increasing number of sound cards to provide

extra audio processing power and help relieve the computers CPU of this

type of work, much like a 3D graphics accelerator would for rendering 3D

graphics

FFT/ SPECTRUM ANALYSER/

HARMONIC ANALYSER / FREQUENCY

MEASUREMENT

The aim of the project is to develop a low cost spectrum analyser

using the DSP processor ADSP. The spectrum can be viewed on the

CRO.

Principle

Any analog signal is a mixture of a number of sinusoidal signals of various

frequencies. The various sinusoidal component frequencies of a signal is

called a spectrum.

A sinusoidal signal is a complex quantity and so it has magnitude and

phase. Hence frequency spectrum consists of magnitude plot and phase

plot. In order to detect the presence of a frequency component, it is

sufficient if we obtain the magnitude spectrum alone.

Hence it is decided to compute the magnitude spectrum using ADSP

processor. The processor performs a FFT operation using radix - 4 FFT

algorithm to determine the magnitude spectrum of the signal.

DSP



digital TVs.

DSP chips are used in sound cards for recording and playback and speech

synthesis. Other audio uses are amplifiers that simulate concert halls and

surround-sound effects for music and home theater







graphics

Hardware

The spectrum analyser system consists of ADSP as CPU, EPROM, RAM,

CODEC and AMPLIFIER. The optional hardware includes serial Input port

and RS 232 level converter for interfacing with PC. Another optional

hardware includes 8031 based waveform generator.

The PC interface can be used to download the object codes of the

program from the PC. The analog signal whose spectrum has to be

estimated is fed to CODEC through MIC or a function generator

(Oscillator).

The CODEC digitize the analog signal and send to ADSP which performs

FFT on the digital signal. The processor output the result of FFT to

CODEC which converts the digital signal to analog. The analog signal is

amplified and displayed on the CRO. The block diagram of the system is

shown in fig.1.

The system can access the analog signal in the frequency range of 0 to

3kHz. The spectrums can be viewed as a voltage spike on the CRO

screen.

For measuring the frequency, a calibration pulse is output by ADSP which

indicates the start of the frequency scale. The divisions on the CRO

screen can be calibrated using a known single frequency signal.

FIR FILTER DESIGN USING ADSP

AIM

The aim of this project is to design FIR filter and to implement the

algorithm using ADSP Processor. The system has four selectable

band pass filters, which is used to modify the voice/music.

HARDWARE

The FIR filter system consists of a DSP microprocessor ADSP

CODEC, EPROM, RAM, Amplifier, Mic and speaker.

The CODEC has been interfaced to ADSP processor through its

serial port.

In this system additional serial port has been developed using buffer

and latch to provide serial communication with PC serial port.

The TTL logic levels of serial port are converted to RS-232 level

using level converter, so that the system can directly communicate

with the standard serial port (com1/com2) of Personal Computer

(PC).

SOFTWARE

The system software and filter algorithm are developed in ADSP

assembly language using PC.

The assembly language programs are compiled using ASM21

assembler and converted to machine code which are either

permanently stored in EPROM or downloaded to the ADSP system

through PC serial interface.

FUNCTIONING

The analog input from the microphone is fed to CODEC. The ADSP

reads the digital signal from CODEC, performs filtering operation,

and then output the processed signal back to CODEC.

The CODEC converts the digital signal back to analog and this

analog signal is fed to the speaker. The desired band pass filter can

be selected by using the interrupt key.

APPLICATIONS

We can implement the following operations on the voice input

1. Filtering the voice on various bands

2. Changing the nature of voice using filtering

ADSP




DSP

DSP stands for Digital Signal Processor. DSP chips are widely used

in sound cards, fax machines, modems, cellular phones, high-

capacity hard disks and digital TVs







processing.





FIR FILTER DESIGN USING ADSP

FUNCTION GENERATOR

The aim of the project is to develop a high precision, function generator

using DSP processor ADSP. The DSP processors are capable of

generating high precision waveforms.

In digital means of generation there is no drift in the generated

waveforms but the drift is inherent in analog systems due to

temperature coefficient of the components.

ADSP


technology which processes the signal to eliminate unwanted noise and

improve incoming signal

DSP


sound cards, fax machines, modems, cellular phones, high-capacity

hard disks and digital TVs


speech synthesis. Other audio uses are amplifiers that simulate concert

halls and surround-sound effects for music and home theater





extra audio processing power and help relieve the computers CPU of

this type of work, much like a 3D graphics accelerator would for

rendering 3D graphics

PRINCIPLE

Any signal to be generated can be represented by a mathematical

equation and it will be a function of the frequency of the desired signal.

For a given frequency, the function can be evaluated at finite number of

points and the signal can be represented as a set of data. These data

can be output to a DAC to generate the required analog signal.

The digital function generator using ADSP has the following

advantages.

1. Precise waveform output.

2. No drift in output due to temperature variations of the

component.

3. Any special type of waveform can be implemented.

HARDWARE

The digital function generator consists of ADSP as CPU, EPROM, RAM,

DAC and Amplifiers. The optional hardware includes PC interface using

serial Input port and level converter. The Input port is implemented

using the buffer

If PC interface is provided, the program for generation of waveform can

be downloaded from PC. When PC interface is not available, the

program can be permanently stored in EPROM.

The program can be executed using the INTERRUPT key. When

interrupt key is pressed the ADSP starts generating the digital waveform

which is output to DAC to convert the digital signal to analog signal.

The analog signal can then be viewed on the CRO.

GRAPHICS ENGINE DESIGN ADSP

ADSP




DSP






concert halls and surround-sound effects for music and home

theater




processing.





The objective of this project is to develop a microcomputer system to

generate various complex geometric models, symbols, images etc.

using line drawings. The system is proposed to be developed using the

DSP microcomputer ADSP.

Any complex symbol or image can be constructed using dots and lines

or curves joining the dots. This forms the basic principle of graphics

generation.

For creation or generation of any object, the object has to be defined by

dots and lines joining the dots in a three-dimensional space consisting of

x, y and z axes. These line data, point data (reference data) are used by

the assembly language program developed using ADSP instruction set

to generate the required object.

HARDWARE

The graphics engine has been designed in this project using ADSP as

CPU. The system consists of ADSP, EPROM, RAM, PC interface,

Buffer, DAC and amplifiers.

The PC interface has been developed using Serial I/O port and RS232

level converter. The output of DACS can be sent to 2 channels of CRO

to view the generated image on the CRO Screen.

FUNCTIONING

The line data and source data is downloaded from the PC to the ADSP

processor.

On pressing the IRQ2 switch on the kit the object that is created using

line data and source data can be viewed in various mode of rotation. The

zoomed object can also be viewed.

Applications

The concept can be used for

1. Video games

2. Ray tracking

3. Three - dimensional animation

4. Shadow casting

5. Hidden line elimination

GRAPHICS ENGINE DESIGN ADSP

IMAGE PROCESSING SYSTEM ADSP

TWO DIMENSIONAL FILTERS

BASICS

Image processing is a visual task. An image consists of two-dimensional

array of numbers. Image processing often involves computation on large

matrices that represents digitized images.

Each element of the array represents a pixel of the image; its location in

the array corresponds to its location in the image and its value

determines the shading of the pixel.

Image processing involves processing or altering an existing image in an

desired manner.

The first step is obtaining an image with image data this is achieved

through a stand-alone application program with CIPS. (C image

processing system).

ADSP




DSP






concert halls and surround-sound effects for music and home

theater




processing.





HARDWARE

This project is proposed to develop an efficient hardware to implement

the “Image Processing” algorithm.

The hardware required for image processing system have been

implemented using powerful DSP processor ADSP.

The image processing systems consists of ADSP, CODEC, EPROM,

RAM serial port and one level converter.

FUNCTIONING

The image data of the image to be processed is downloaded from the

PC to the ADSP.

The image data from the PC is processed in the ADSP processor (either

filtering or edge detection process) and than the kit on reset, the

processed image can be viewed on the VDU.

APPLICATIONS

Image processing is popular in applications like

1. Remote sensing via satellites

2. Radar

3. Sonar

IMAGE PROCESSING SYSTEM ADSP

TWO DIMENSIONAL FILTERS

NOISE/ ECHO CANCELLOR NOISE/ ECHO CANCELLOR NOISE/ ECHO CANCELLOR NOISE/ ECHO CANCELLOR

AIM

The aim of the project is to develop an adaptive filter to eliminate

echo/noise .The adaptive filter is implemented in ADSP based system.

THEORY

Normally, in FIR filter, the filter coefficients are constant. But in the

adaptive FIR filters, the filter coefficients will be adjusted or adapt itself

to minimize the errors.

Thus when we employ adaptive filter for echo/noise cancellation, the

FIR filter coefficients are adjusted in every cycle to remove the

echo/noise from the input. In this project Least Mean Square (LMS)

algorithm is used for adaptive filter.

HARDWARE

The physical system consists of a DSP microprocessor ADSP, CODEC,

EPROM, RAM, Amplifiers, Mic, and Speaker. The CODEC has been

interfaced to ADSP processor through its serial port.

In this system additional serial port has been developed using the buffer

and latch to provide serial communication with PC serial port.

The TTL logic levels of serial port are converted to RS-232 logic levels

using converters so that the system can directly communicate with the

standard serial port (com1/com2) of the Personal Computer (PC).

SOFTWARE

The system software and the adaptive filter LMS algorithm are

developed in ADSP assembly language using PC.

The Assembly language programs are compiled using ASM21

Assembler and converted to machine code which are either

permanently stored in EPROM or downloaded to the ADSP system

through PC serial interface.

FUNCTIONING

The analog input signal from the mic are amplified and fed to the

CODEC. The CODEC converts the analog signal into digital and

transfers the data to DSP microprocessor.

In order to demonstrate echo/noise cancellation, first processor

generates an echo/noise and mix with the input signal and the resultant

signal is output to speaker.

Then using the interrupt adaptive filter algorithm is enabled and the

combined signal is processed by the adaptive filter and the filtered

signal is output through speaker. We can observe the gradual

elimination of echo/noise.

APPLICATIONS

The Adaptive filter is widely used in applications like

1. Digital audio

2. Digital graphical equalizer

3. Noise removal in transmission channels

4. High speed modem

5. Echo cancellation in transmission channels.

ADSP



incoming signal

DSP



digital TVs. DSP chips are used in sound cards for recording and playback

and speech synthesis. Other audio uses are amplifiers that simulate concert

halls and surround-sound effects for music and home theater

Digital Signal Processing uses mathmatics to operate on a digital signal (such

as a digital audio stream) to generate some type of altered output. DSP is

used heavily in software and hardware effects processing.

DSP chips are found on an increasing number of sound cards to provide extra

audio processing power and help relieve the computers CPU of this type of

work, much like a 3D graphics accelerator would for rendering 3D graphics

NOISE/ ECHO CANCELLOR NOISE/ ECHO CANCELLOR NOISE/ ECHO CANCELLOR NOISE/ ECHO CANCELLOR

SPEECH COMPRESSION AND SPEECH COMPRESSION AND SPEECH COMPRESSION AND SPEECH COMPRESSION AND

DECOMPRESSION USING ADPCM DECOMPRESSION USING ADPCM DECOMPRESSION USING ADPCM DECOMPRESSION USING ADPCM

TECHNIQUETECHNIQUETECHNIQUETECHNIQUE

BASICS

Speech coding may be defined as a digital representation of the

speech signal that provides efficient storage, transmission, recovery

and faithful reconstruction of the original speech.

The speech compression is achieved by representing each sample of

digitized data by lesser number of bits. For maintaining the voice

quality the speech sample has to be represented by 13 to 16 bits.

In compression techniques the 13 to 16 bits data is represented by

an equivalent 4 or 8 bit data. In decompression techniques the

compressed data is converted back to original data. One of the

important requirement in these techniques is that the voice quality

has to be maintained.

HARDWARE

This project is proposed to develop an efficient hardware to

implement the powerful and versatile ADPCM speech compression

and decompression algorithm.

The hardware required for speech compression and decompression

have been developed using powerful DSP processor ADSP-2105.

The speech processing system consists of ADSP - 2105, CODEC,

EPROM, RAM, Amplifier sections, mike and speakers.

FUNCTIONING

The system is ready to accept speech signal once it is reset. The

speech segment to be compressed is fed as input through

microphone. The analog speech signal is digitized by CODEC,

compressed by ADSP-2105 and stored in RAM.

The decompression process can be started by interrupting the

processor.

When the processor is interrupted, the information in memory is

fetched by the processor and decompressed and is then fed to

CODEC. The CODEC converts the digital signal to analog signal.

The analog signal is amplified and output through the speaker.

FEATURES

The speech compression and decompression techniques are

implemented in cellular phones, voice mail transmission systems and

speech recognition system.

What is ADPCM

Adaptive delta pulse code modulation. An audio compression

algorithm for digital audio based on describing level differences

between adjacent samples

Adaptive Differential Pulse Code Modulation is an audio encoding

compression technique that encodes the difference between the

predicted value of the signal instead of the absolute value of the

original waveform so that the compression efficiency is improved.

This difference is usually small and can thus be encoded in fewer bits

than the sample itself. ADPCM is used in PC sound cards, on CD-i

and CD-ROM XA discs

CODEC

Codec is an abbreviation for Coder-Decoder. It's an analog-to-digital

(A/D) and digital-to-analog (D/A) converter for translating the signals

from the outside world to digital, and back again

ADSP




DSP










processing.





VOICE COMPERSION USING ADSP

What is Voice Compression

The conversion of an analog voice signal into a digital signal using minimum

bandwidth, Process by which the high statistical correlation between

consecutive voice samples is used to create a variable quantizing scale.

ADPCM can encode analog voice samples into high - toll - quality digital

signals

DSP Processor are widely used for analog signal processing such as

filtering, compression of voice signal and image processing. The processor

has powerful instruction sets and most of the instructions are one cycle

execution. So that the result can be obtained much faster and efficient,

which is required for Analog signal processing.

In this project, we are using ADSP (DSP) processor for voice

compression and retrieving.

This voice signal is picked up by microphone and it is amplified by the Pre-

Amplifier to 0-5V level. The amplified signal fed to CODEC. The CODEC has

an ADC (Analog Digital Converter), which receives the signal from pre-amp

and converts it in to Digital data.

The Converted digital data is encoded in serial format and transmitted to

DSP processor by the CODEC. The CODEC and DSP processor are linked

by synchronized serial port.

The processor has two memory blocks externally, Data memory and Boot

memory. On power on reset the program from the Boot memory (which is

developed and stored in EPROM) is loaded to the internal Program memory

of the processor. After the completing loading sequence it starts executing.

The program is developed such that to receive the data from CODEC and

compress it using Linear Predictive Coding (LPC).

The compressed data are stored in external data memory (DM). When the

IRQ pin is asserted the data from DM is read and expanded. The expanded

data is transmitted to the serial Port (to CODEC)

The CODEC receives the data and decodes it. The decoded data is fed to

DAC (which is build-in CODEC). And the DAC reproduces the exact analog

signal and amplified by the Audio Power Amplifier

What is ADPCM

Adaptive delta pulse code modulation. An audio compression algorithm for

digital audio based on describing level differences between adjacent

samples

Adaptive Differential Pulse Code Modulation is an audio encoding

compression technique that encodes the difference between the predicted

value of the signal instead of the absolute value of the original waveform so

that the compression efficiency is improved.

This difference is usually small and can thus be encoded in fewer bits than

the sample itself. ADPCM is used in PC sound cards, on CD-i and CD-ROM

XA discs

CODEC

Codec is an abbreviation for Coder-Decoder. It's an analog-to-digital (A/D)

and digital-to-analog (D/A) converter for translating the signals from the

outside world to digital, and back again

ADSP



incoming signal

DSP


sound cards, fax machines, modems, cellular phones, high-capacity hard

disks and digital TVs

DSP chips are used in sound cards for recording and playback and speech

synthesis. Other audio uses are amplifiers that simulate concert halls and

surround-sound effects for music and home theater







graphics

The block diagram is shown below.

DATA

MEMORY

BOOT

MEMORY

CODEC

ADSP

2105

DSP

PRE

AMP

MIC

PWR

AMP

SPK

Technology

Dsp