DIGITAL SIGNAL PROCESSING

At sivagangai
Presented by

PRABAKARAN.M (ECE IIND YEAR) KARTHICK KANDASAMY.R (ECE IIND YEAR)

E-MAIL ID:

mpkaran89@yahoo.co.in vista_kk@yahoo.co.in

CONTENTS

Digital signal processor

Characteristics of DSP Analog & digital processing Signal processing Speech processing Information processing Analog &signal processing ADC DAC DSPs in 2007 Applications

ABSTRACT
The DSP is the study of signals in a digital representation and the processing methods of these signals. The characteristics tell about analog signals and digital signals, signal processing, speech processing, information processing In other hands DSP tells about ADC and DACthe development of 2007 the DSP and applications that are used by us in domestic life..

OPENING REMARKS
Digital signal processing (DSP):
It is the study of signals in a digital representation and the processing methods of these signals. DSP and analog signal processing are subfields of signal processing. DSP includes subfields like: audio and speech signal processing, sonar and radar signal processing, sensor array processing, spectral estimation, statistical signal processing, image processing, signal processing for communications, biomedical signal processing, etc. Since the goal of DSP is usually to measure or filter continuous real-world analog signals, the first step is usually to convert the signal from an analog to a digital form, by using an analog to digital converter. Often, the required output signal is another analog output signal, which requires a digital to analog converter. The algorithms required for DSP are sometimes performed using specialized computers, which make use of specialized microprocessors called digital signal processors (also abbreviated DSP).

Digital signal processor (DSP):

It is a specialized microprocessor designed specifically for digital signal processing, generally in real-time computing.

Microprocessor

A microprocessor is a programmable digital electronic component that incorporates the functions of a central processing unit (CPU) on a single semi conducting integrated circuit (IC). The microprocessor was born by reducing the word size of the CPU from 32 bits to 4 bits, so that the transistors of its logic circuits would fit onto a single part. One or more microprocessors typically serve as the CPU in a computer system, embedded system, or handheld device. Microprocessors made possible the advent of the microcomputer in the mid1970s. Before this period, electronic CPUs were typically made from bulky discrete switching devices (and later small-scale integrated circuits) containing the equivalent of only a few transistors.

Characteristics of typical Digital Signal Processors

Designed for real-time processing Optimum performance with streaming data Separate program and data memories (Harvard architecture) Special Instructions for SIMD (Single Instruction, Multiple Data) operations No hardware support for multitasking The ability to act as a direct memory access device if in a host environment Processes digital signals converted (using an Analog-to-digital converter (ADC)) from analog signals. Output is then converted back to analog form using a Digital-to-analog converter (DAC)

Block diagram

Analog and digital signals

In many cases, the signal of interest is initially in the form of an analog electrical voltage or current, produced for example by a microphone or some other type of transducer. In some situations, such as the output from the readout system of a CD (compact disc) player, the data is already in digital form. An analog signal must be converted into digital form before DSP techniques can be applied. An analog electrical voltage signal, for example, can be digitized using an electronic circuit called an analog-to-digital converter or ADC. This generates a digital output as a stream of binary numbers whose values represent the electrical voltage input to the device at each sampling instant.

Signal processing
Signal processing is the analysis, interpretation and manipulation of signals. Signals of interest include sound, images, biological signals such as ECG, radar signals, and many others. Processing of such signals includes storage and reconstruction, separation of information from noise (e.g., aircraft identification by radar), compression (e.g., Signal classification There are various sorts of signal processing, depending on the nature of the signal, as in the following examples. For analog signals, signal processing may involve the amplification and filtering of audio signals for audio equipment or the modulation and demodulation of signals for telecommunications. For digital signals, signal processing may involve the compression, error checking and error detection of digital signals. Analog signal processingfor signals that have not been digitized, as in classical radio, telephone, radar, and television systems Digital signal processingfor signals that have been digitized. Processing is done by digital circuits such as ASICs, FPGAs, general-purpose microprocessors or computers, or specialized digital signal processor chips. Statistical signal processinganalyzing and extracting information from signals based on their statistical properties Audio signal processingfor electrical signals representing sound, such as music Speech signal processingfor processing and interpreting spoken words Image processingin digital cameras, computers, and various imaging systems Video signal processingfor interpreting moving pictures Array processingfor processing signals from arrays of sensors

Speech signal processing

Speech signal processing refers to the acquisition, manipulation, storage, transfer and output of human utterances by a computer. The main goals are the recognition, synthesis and compression of human speech:

Speech recognition (also called voice recognition) focuses on capturing the human voice as a digital sound wave and converting it into a computerreadable format.

Speech synthesis is the reverse process of speech recognition. Advances in this area improve the computers' usability for the visually impaired. Speech compression is important in the telecommunications area for increasing the amount of info which can be transferred, stored, or heard, for a given set of time and space constraints.

Multilingual Speech Processing, Edited by Tanja Schultz and Katrin Kirchhoff, April 2006--Researchers and developers in industry and academia with different backgrounds but a common interest in multilingual speech processing will find an excellent overview of research problems and solutions detailed from theoretical and practical perspectives.---CH 1: Introduction / CH 2: Language Characteristics / CH 3: Linguistic Data Resources / CH 4:

Multilingual Acoustic Modeling / CH 5: Multilingual Dictionaries / CH 6: Multilingual Language Modeling / CH 7: Multilingual Speech Synthesis / CH 8: Automatic Language Identification / CH 9: Other Challenges / CH 10: Speech-to-Speech Translation / CH 11: Multilingual Spoken Dialog Systems / Bibliography

Information processing
Information processing is the changing (processing) of information in any manner detectable by an observer. As such, it is a process which describes everything which happens (changes) in the universe, from the falling of a rock (a change in position) to the printing of a text file from a digital computer system. In the latter case, an information processor is changing the form of presentation of that text file. Information processing may more specifically be defined in terms by Claude E. Shannon as the conversion of latent information into manifest information Latent and manifest information is defined through the terms of equivocation (remaining uncertainty, what value the sender has actually chosen), dissipation (uncertainty of the sender what the receiver has actually received) and transformation (saved effort of questioning - equivocation minus dissipation)

Analog signal processing

Analog signal processing is any signal processing conducted on analog signals by analog means. Specifically the mathematical algorithm that processes the signal is implemented with analog electronics in which the mathematical values are represented as a continuous physical quantity, an analog, usually as a voltage,

electric current, or electric charge around some components in the electronic devices. A small error or noise affecting such physical quantities will result in a corresponding error in the signals represented by such physical quantites.

Analog-to-digital converter

An analog-to-digital converter (abbreviated ADC, A/D or A to D) is an electronic integrated circuit (i/c) that converts continuous signals to discrete digital numbers. The reverse operation is performed by a digital-to-analog converter (DAC). Typically, an ADC is an electronic device that converts an input analog voltage (or current) to a digital number. The digital output may be using different coding schemes, such as binary and two's complement binary. However, some nonelectronic or only partially electronic devices, such as rotary encoders, can also be considered ADCs. The resolution of the converter indicates the number of discrete values it can produce over the range of analog values. The values are usually stored electronically in binary form, so the resolution is usually expressed in bits. In consequence, the number of discrete values available, or "levels", is usually a power of two. For example, an ADC with a resolution of 8 bits can encode an analog input to one in 256 different levels, since 28 = 256. The values can represent the ranges 0 to 255 or -128 to 127, for example, depending on the application.

Resolution can also be defined electrically, and expressed in volts. The voltage resolution of an ADC is equal to its overall voltage measurement range divided by the number of discrete intervals as in the formula:

Where Q is resolution in volts, EFSR is the full scale voltage range, and M is resolution in bits. The number of intervals is given by the number of available levels minus one. In practice, the useful resolution of the converter is limited by the signal-tonoise ratio of the signal in question. If there is too much noise present in the analog input, it will be impossible to accurately resolve beyond a certain number of bits of resolution, the "effective number of bits" (ENOB). While the ADC will produce a result, the result is not accurate, since its lower bits are simply measuring noise. The signal-to-noise ratio should be around 6 dB per bit of resolution required.

Response type
Linear ADCs
Most ADCs are of a type known as linear, although analog-to-digital conversion is an inherently non-linear process (since the mapping of a continuous space to a discrete space is a non-invertible and therefore non-linear operation). The term linear as used here means that the range of the input values that map to each output value has a linear relationship with the output value, i.e., that the output value k is used for the range of input values from M (k + b) To M (k + 1 + b),

Where m and b are constants. Here b is typically 0 or 0.5. When b = 0, the ADC is referred to as mid-rise, and when b = 0.5 it is referred to as mid-tread.

Non-linear ADCs
If the probability density function of a signal being digitized is uniform, then the signal-to-noise ratio relative to the quantization noise is the best possible. Because of this, it's usual to pass the signal through its cumulative distribution function (CDF) before the quantization. This is good because the regions that are more important get quantized with a better resolution. In the dequantization process, the inverse CDF is needed. This is the same principle behind the companders used in some taperecorders and other communication systems, and is related to entropy maximization. (Never confuse companders with compressors!) For example, a voice signal has a Laplacian distribution. This means that the region around the lowest levels, near 0, carries more information than the regions with higher amplitudes. Because of this, logarithmic ADCs are very common in voice communication systems to increase the dynamic range of the representable values while retaining fine-granular fidelity in the low-amplitude region. An eight-bit a-law or the -law logarithmic ADC covers the wide dynamic range and has a high resolution in the critical low-amplitude region that would otherwise require a 12-bit linear ADC.

Accuracy
An ADC has several sources of errors. Quantization error and (assuming the ADC is intended to be linear) non-linearity is intrinsic to any analog-to-digital conversion. There is also a so-called aperture error which is due to a clock jitter and is revealed when digitizing a signal (not a single value).

These errors are measured in a unit called the LSB, which is an abbreviation for least significant bit. In the above example of an eight-bit ADC, an error of one LSB is 1/256 of the full signal range, or about 0.4%.

Commercial analog-to-digital converters

Most converters sample with 6 to 24 bits of resolution, and produce fewer than 1 mega sample per second. It is rare to get more than 24 bits of resolution. Mega- and gig sample converters are available, though (Feb 2002). Megasample converters are required in digital video cameras, video capture cards, and TV tuner cards to convert full-speed analog video to digital video files. Commercial converters usually have 0.5 to 1.5 LSB error in their output. In many cases the most expensive part of an integrated circuit is the pins, because they make the package larger, and each pin has to be connected to the integrated circuit's silicon. To save pins, it's common for slow ADCs to send their data one bit at a time over a serial interface to the computer, with the next bit coming out when a clock signal changes state, say from zero to 5V. This saves quite a few pins on the ADC package, and in many cases, does not make the overall design any more complexes. (Even microprocessors which use memory-mapped IO only need a few bits of a port to implement a serial bus to an ADC.) Commercial ADCs often have several inputs that feed the same converter, usually through an analog multiplexer. Different models of ADC may include sample and hold circuits, instrumentation amplifiers or differential inputs, where the quantity measured is the difference between two voltages

Digital-to-analog converter

In electronics, a digital-to-analog converter (DAC or D-to-A) is a device for converting a digital (usually binary) code to an analog signal (current, voltage or electric charge). Digital-to-analog converters are interfaces between the abstract digital world and analog real life. An analog-to-digital converter (ADC) performs the reverse operation. DAC usually only deals with pulse-code modulation (PCM)-encoded signals. The job of converting various compressed forms of signals into PCM is left to codecs.

Basic ideal operation

The DAC fundamentally converts finite-precision numbers (usually fixedpoint binary numbers) into a physical quantity, usually an electrical voltage. Normally the output voltage is a linear function of the input number. Usually these numbers are updated at uniform sampling intervals and can be thought of as numbers obtained from a sampling process. These numbers are written to the DAC, sometimes along with a clock signal that causes each number to be latched in

sequence, at which time the DAC output voltage changes rapidly from the previous value to the value represented by the currently latched number. The effect of this is that the output voltage is held in time at the current value until the next input number is latched resulting in a piecewise constant output. This is equivalently a zero-order hold operation and has an effect on the frequency response of the reconstructed signal.

Piecewise constant signal typical of a practical DAC output. The fact that practical DACs do not output a sequence of dirac impulses (that, if ideally low-pass filtered, result in the original signal before sampling) but instead output a sequence of piecewise constant values or rectangular pulses, means that there is an inherent effect of the zero-order hold on the effective frequency response of the DAC resulting in a mild roll-off of gain at the higher frequencies (a 3.9224 dB loss at the Nyquist frequency). This zero-order hold effect is a consequence of the hold action of the DAC and is not due to the sample and hold that might precede a conventional analog to digital converter as is often misunderstood.

DAC types
The most common types of electronic DACs are:

The Pulse Width Modulator, the simplest DAC type. A stable current or voltage is switched into a low pass analog filter with a duration determined by the digital input code. This technique is often used for electric motor speed control, and is now becoming common in high-fidelity audio. Oversampling DACs such as the Delta-Sigma DAC, use a pulse density conversion technique. The oversampling technique allows for the use of a lower resolution DAC internally. A simple 1-bit DAC is often chosen because the oversampled result is inherently linear. The DAC is driven with a pulse density modulated signal, created with the use of a low-pass filter, step nonlinearity (the actual 1bit DAC), and negative feedback loop, in a technique called delta-sigma modulation. This results in an effective high-pass filter acting on the quantization (signal processing) noise, thus steering this noise out of the low frequencies of interest into the high frequencies of little interest, which is called noise shaping (very high frequencies because of the oversampling). The quantization noise at these high frequencies are removed or greatly attenuated by use of an analog low-pass filter at the output (sometimes a simple RC lowpass circuit is sufficient). Most very high resolution DACs (greater than 16 bits) is of this type due to its high linearity and low cost. Higher oversampling rates can either relax the specifications of the output low-pass filter or enable further suppression of quantization noise. Speeds of greater than 100 thousand samples per second (for example, 192kHz) and resolutions of 24 bits are attainable with Delta-Sigma DACs. A short comparison with pulse width modulation shows that an 1-bit DAC with a simple first-order integrator would have to run at 3 THz (which is physically unrealizable) to archive 24 meaningful bits of resolution, requiring a higher order low-pass filter in the noise-shaping loop. A single integrator is a low pass filter with a frequency response inversely proportional to frequency and using one such integrator in the noise-shaping loop is a first order delta-sigma modulator. Multiple higher order topologies (such as MASH) are used to achieve higher degrees of noiseshaping with a stable topology.

The Binary Weighted DAC, which contains one resistor or current source for each bit of the DAC connected to a summing point. These precise voltages or currents sum to the correct output value. This is one of the fastest conversion methods but suffers from poor accuracy because of the high precision required for each individual voltage or current. Such high-precision resistors and current-sources are expensive, so this type of converter is usually limited to 8bit resolution or less. The R-2R Ladder DAC, which is a binary weighted DAC that uses a repeating cascaded structure of resistor values R and 2R. This improves the precision due to the relative ease of producing equal valued matched resistors (or current sources). However, wide converters perform slowly due to increasingly large RC-constants for each added R-2R link. The Thermometer coded DAC, which contains an equal resistor or current source segment for each possible value of DAC output. An 8-bit thermometer DAC would have 255 segments, and a 16-bit thermometer DAC would have 65,535 segments. This is perhaps the fastest and highest precision DAC architecture but at the expense of high cost. Conversion speeds of >1 billion samples per second have been reached with this type of DAC. The Segmented DAC, which combines the thermometer coded principle for the most significant bits and the binary weighted principle for the least significant bits. In this way, a compromise is obtained between precision (by the use of the thermometer coded principle) and number of resistors or current sources (by the use of the binary weighted principle). The full binary weighted design means 0% segmentation; the full thermometer coded design means 100% segmentation. Hybrid DACs, which use a combination of the above techniques in a single converter. Most DAC integrated circuits are of this type due to the difficulty of getting low cost, high speed and high precision in one device.

DAC performance
DACs are at the beginning of the analog signal chain, which makes them very important to system performance. The most important characteristics of these devices are:

Resolution: This is the number of possible output levels the DAC is

designed to reproduce. This is usually stated as the number of bits it uses, which is the base two logarithm of the number of levels. For instance a 1 bit DAC is designed to reproduce 2 (21) levels while an 8 bit DAC is designed for 256 (28) levels. Resolution is related to the Effective Number of Bits (ENOB) which is a measurement of the actual resolution attained by the DAC.

Maximum sampling frequency: This is a measurement of the maximum

speed at which the DACs circuitry can operate and still produce the correct output. As stated in the Shannon-Nyquist sampling theorem, a signal must be sampled at over twice the bandwidth of the desired signal. For instance, to reproduce signals in all the audible spectrum, which includes frequencies of up to 20 kHz, it is necessary to use DACs that operate at over 40 kHz. The CD standard samples audio at 44.1 kHz, thus DACs of this frequency are often used. A common frequency in cheap computer sound cards is 48 kHz many work at only this frequency, offering the use of other sample rates only through (often poor) internal resampling.

Monotonicity: This refers to the ability of DACs analog output to increase

with an increase in digital code or the converse. This characteristic is very important for DACs used as a low frequency signal source or as a digitally programmable trim element.

THD+N: This is a measurement of the distortion and noise introduced to

the signal by the DAC. It is expressed as a percentage of the total power of unwanted harmonic distortion and noise that accompany the desired signal.

This is a very important DAC characteristic for dynamic and small signal DAC applications.

Dynamic range: This is a measurement of the difference between the

largest and smallest signals the DAC can reproduce expressed in Decibels. This is usually related to DAC resolution and noise floor.

Other measurements, such as Phase distortion and Sampling Period Instability, can also be very important for some applications.

DAC figures of merit

Static performance:
DNL (Differential Non-Linearity) shows how much two adjacent code analog values deviate from the ideal 1LSB step [1] INL (Integrated Non-Linearity) shows how much the DAC transfer characteristic deviates from an ideal one. That is, the ideal characteristic is usually a straight line; INL shows how much the actual voltage at a given code value differs from that line, in LSBs (1LSB steps). Gain

DSPs in 2007
Todays signal processors yield much greater performance. This is due in part to both technological and architectural advancements like lower design rules, fast-access two-level cache, (E) DMA circuit and a wider bus system. Of course, not all DSPs provide the same speed and many kinds of signal processors exist, each one of them being better suited for a specific task, ranging in price from about US$1.50 to US$300. A Texas Instruments C6000 series DSP clocks at 1 GHz and implements separate instruction and data caches as well as a 8 MiB 2nd level cache, and its I/O

speed is rapid thanks to its 64 EDMA channels. The top models are capable of even 8000 MIPS (million instructions per second), use VLIW encoding, perform eight operations per clock-cycle and are compatible with a broad range of external peripherals and various buses (PCI/serial/etc). Another big signal processor manufacturer today is Analog Devices. The company provides a broad range of DSPs, but its main portfolio is multimedia processors, such as codecs, filters and digital-analog converters. Its SHARC-based processors range in performance from 66 MHz/198 MFLOPS (million floating-point operations per second) to 400 MHz/2400MFLOPS. Some models even support multiple multipliers and ALUs, SIMD instructions and audio processing-specific components and peripherals. Another product of the company is the Blackfin family of embedded digital signal processors, with models like the ADSP-BF531 to ADSPBF536. These processors combine the features of a DSP with those of a general use processor. As a result, these processors can run simple operating systems like CLinux, velOSity and Nucleus RTOS while operating relatively efficiently on realtime data.

Applications
The main applications of DSP are audio signal processing, audio compression, digital image processing, video compression, speech processing, speech recognition, digital communications, RADAR, SONAR, seismology, and biomedicine. Specific examples are speech compression and transmission in digital mobile phones, room matching equalisation of sound in Hifi and sound

reinforcement applications, weather forecasting, economic forecasting, seismic data processing, analysis and control of industrial processes, computer-generated animations in movies, medical imaging such as CAT scans and MRI, image manipulation, high fidelity loudspeaker crossovers and equalization, and audio effects for use with electric guitar amplifiers.

TERMINATION

Thus DSP playing an important rolling in the electronic system Thus DSP playing important role in the satellite communication. The DSP is morely used in medical field. The DSP tells the characteristics of signals & the conversation purpose. Thus the signals are easily filtered &the shape of the amplitude canbe easily changed by this process.