DOCSLIB.ORG

Digital Audio Compression

Total Page：16

File Type：pdf, Size：1020Kb

Download full-text PDF Read full-text
Digital Audio Compression Digital Audio Compression By Davis Yen Pan begins with a summary of the basic audio digitiza- Abstract tion process. The next two sections present detailed descriptions of two relatively simple approaches to Compared to most digital data types, with the audio compression: "-law and adaptive differential exception of digital video, the data rates associ- pulse code modulation. In the following section, the ated with uncompressed digital audio are substan- paper gives an overview of a third, much more so- tial. Digital audio compression enables more effi- phisticated, compression audio algorithm from the cient storage and transmission of audio data. The Motion Picture Experts Group. The topics covered many forms of audio compression techniques offer a in this section are quite complex and are intended range of encoder and decoder complexity, compressed for the reader who is familiar with digital signal audio quality, and differing amounts of data com- processing. The paper concludes with a discussion pression. The "-law transformation and ADPCM of software-only real-time implementations. coder are simple approaches with low-complexity, low-compression, and medium audio quality algo- Digital Audio Data rithms. The MPEG/audio standard is a high- complexity, high-compression, and high audio qual- The digital representation of audio data offers ity algorithm. These techniques apply to general au- many advantages: high noise immunity, stability, dio signals and are not specifically tuned for speech and reproducibility. Audio in digital form also al- signals. lows the efficient implementation of many audio processing functions (e.g., mixing, filtering, and equalization) through the digital computer. Introduction The conversion from the analog to the digital do- Digital audio compression allows the efficient stor- main begins by sampling the audio input in regular, age and transmission of audio data. The various au- discrete intervals of time and quantizing the sam- dio compression techniques offer different levels of pled values into a discrete number of evenly spaced complexity, compressed audio quality, and amount levels. The digital audio data consists of a sequence of data compression. of binary values representing the number of quan- This paper is a survey of techniques used to com- tizer levels for each audio sample. The method of press digital audio signals. Its intent is to provide representing each sample with an independent code useful information for readers of all levels of ex- word is called pulse code modulation (PCM). Figure perience with digital audio processing. The paper 1 shows the digital audio process. 00110111000... 11001100100... ANALOG ANALOG AUDIO PCM PCM AUDIO INPUT VALUES VALUES OUTPUT ANALOG-TO-DIGITAL DIGITAL SIGNAL DIGITAL-TO-ANALOG CONVERSION PROCESSING CONVERSION Figure 1 Digital Audio Process Digital Technical Journal Vol. 5 No. 2, Spring 1993 1 Digital Audio Compression According to the Nyquist theory, a time-sampled signals.[3,4] The federal standards 1015 LPC (linear signal can faithfully represent signals up to half the predictive coding) and 1016 CELP (coded excited lin- sampling rate.[1] Typical sampling rates range from ear prediction) fall into this category of audio com- 8 kilohertz (kHz) to 48 kHz. The 8-kHz rate covers pression. a frequency range up to 4 kHz and so covers most of the frequencies produced by the human voice. The 48-kHz rate covers a frequency range up to 24 kHz "-law Audio Compression and more than adequately covers the entire audi- " ble frequency range, which for humans typically ex- The -law transformation is a basic audio compres- tends to only 20 kHz. In practice, the frequency sion technique specified by the Comité Consultatif range is somewhat less than half the sampling rate Internationale de Télégraphique et Téléphonique because of the practical system limitations. (CCITT) Recommendation G.711.[5] The transfor- mation is essentially logarithmic in nature and al- The number of quantizer levels is typically a power lows the 8 bits per sample output codes to cover of 2 to make full use of a fixed number of bits per au- a dynamic range equivalent to 14 bits of linearly dio sample to represent the quantized values. With quantized values. This transformation offers a com- uniform quantizer step spacing, each additional bit pression ratio of (number of bits per source sample) has the potential of increasing the signal-to-noise /8 to 1. Unlike linear quantization, the logarithmic ratio, or equivalently the dynamic range, of the step spacings represent low-amplitude audio sam- quantized amplitude by roughly 6 decibels (dB). The ples with greater accuracy than higher-amplitude typical number of bits per sample used for digital values. Thus the signal-to-noise ratio of the trans- audio ranges from 8 to 16. The dynamic range ca- formed output is more uniform over the range of pability of these representations thus ranges from amplitudes of the input signal. The "-law transfor- 48 to 96 dB, respectively. To put these ranges into mation is perspective, if 0 dB represents the weakest audi- & ' ble sound pressure level, then 25 dB is the mini- PSS 127 ln@I C "jxjA for x ! H y a ln(1+") mum noise level in a typical recording studio, 35 dB IPU 127 ln@I C "jxjA for x ` H ln(1+") is the noise level inside a quiet home, and 120 dB is the loudest level before discomfort begins.[2] In terms of audio perception, 1 dB is the minimum au- dible change in sound pressure level under the best where m = 255, and x is the value of the input sig- conditions, and doubling the sound pressure level nal normalized to have a maximum value of 1. The amounts to one perceptual step in loudness. CCITT Recommendation G.711 also specifies a simi- lar A-law transformation. The "-law transformation Compared to most digital data types (digital video is in common use in North America and Japan for excluded), the data rates associated with uncom- the Integrated Services Digital Network (ISDN) 8- pressed digital audio are substantial. For example, kHz-sampled, voice-grade, digital telephony service, the audio data on a compact disc (2 channels of au- and the A-law transformation is used elsewhere for dio sampled at 44.1 kHz with 16 bits per sample) re- the ISDN telephony. quires a data rate of about 1.4 megabits per second. There is a clear need for some form of compression to enable the more efficient storage and transmis- Adaptive Differential Pulse Code sion of this data. Modulation The many forms of audio compression techniques Figure 2 shows a simplified block diagram of an differ in the trade-offs between encoder and decoder adaptive differential pulse code modulation (AD- complexity, the compressed audio quality, and the PCM) coder.[6] For the sake of clarity, the figure amount of data compression. The techniques pre- omits details such as bit-stream formatting, the pos- sented in the following sections of this paper cover sible use of side information, and the adaptation the full range from the "-law, a low-complexity, blocks. The ADPCM coder takes advantage of the low-compression, and medium audio quality algo- fact that neighboring audio samples are generally rithm, to MPEG/audio, a high-complexity, high- similar to each other. Instead of representing each compression, and high audio quality algorithm. audio sample independently as in PCM, an ADPCM These techniques apply to general audio signals and encoder computes the difference between each au- are not specifically tuned for speech signals. This dio sample and its predicted value and outputs the paper does not cover audio compression algorithms PCM value of the differential. Note that the AD- designed specifically for speech signals. These al- PCM encoder (Figure 2a) uses most of the compo- gorithms are generally based on a modeling of the nents of the ADPCM decoder (Figure 2b) to compute vocal tract and do not work well for nonspeech audio the predicted values. 2 Digital Technical Journal Vol. 5 No. 2, Spring 1993 Digital Audio Compression The following section describes the ADPCM algo- rithm proposed by the Interactive Multimedia Asso- ciation (IMA). This algorithm offers a compression X[n] + D[n] C[n] + (ADAPTIVE) factor of (number of bits per source sample)/4 to 1. QUANTIZER – Other ADPCM audio compression schemes include the CCITT Recommendation G.721 (32 kilobits per second compressed data rate) and Recommendation Xp[n – 1] (ADAPTIVE) (ADAPTIVE) G.723 (24 kilobits per second compressed data rate) PREDICTOR DEQUANTIZER Xp[n] standards and the compact disc interactive audio + Dq[n] compression algorithm.[7,8] + + The IMA ADPCM Algorithm. The IMA is a consor- tium of computer hardware and software vendors cooperating to develop a de facto standard for com- (a) ADPCM Encoder puter multimedia data. The IMA’s goal for its audio compression proposal was to select a public-domain audio compression algorithm able to provide good C[n] Dq[n] + Xp[n] compressed audio quality with good data compres- (ADAPTIVE) + DEQUANTIZER sion performance. In addition, the algorithm had to + be simple enough to enable software-only, real-time Xp[n – 1] (ADAPTIVE) decompression of stereo, 44.1-kHz-sampled, audio PREDICTOR signals on a 20-megahertz (MHz) 386-class com- puter. The selected ADPCM algorithm not only meets these goals, but is also simple enough to en- (b) ADPCM Decoder able software-only, real-time encoding on the same computer. Figure 2 ADPCM Compression and Decompression The simplicity of the IMA ADPCM proposal lies in the crudity of its predictor. The predicted value of the audio sample is simply the decoded value of the immediately previous audio sample. Thus the The quantizer output is generally only a (signed) predictor block in Figure 2 is merely a time-delay representation of the number of quantizer levels.
Load more

Recommended publications
  • UC Riverside UC Riverside Electronic Theses and Dissertations
    UC Riverside UC Riverside Electronic Theses and Dissertations Title Sonic Retro-Futures: Musical Nostalgia as Revolution in Post-1960s American Literature, Film and Technoculture Permalink https://escholarship.org/uc/item/65f2825x Author Young, Mark Thomas Publication Date 2015 Peer reviewed|Thesis/dissertation eScholarship.org Powered by the California Digital Library University of California UNIVERSITY OF CALIFORNIA RIVERSIDE Sonic Retro-Futures: Musical Nostalgia as Revolution in Post-1960s American Literature, Film and Technoculture A Dissertation submitted in partial satisfaction of the requirements for the degree of Doctor of Philosophy in English by Mark Thomas Young June 2015 Dissertation Committee: Dr. Sherryl Vint, Chairperson Dr. Steven Gould Axelrod Dr. Tom Lutz Copyright by Mark Thomas Young 2015 The Dissertation of Mark Thomas Young is approved: Committee Chairperson University of California, Riverside ACKNOWLEDGEMENTS As there are many midwives to an “individual” success, I’d like to thank the various mentors, colleagues, organizations, friends, and family members who have supported me through the stages of conception, drafting, revision, and completion of this project. Perhaps the most important influences on my early thinking about this topic came from Paweł Frelik and Larry McCaffery, with whom I shared a rousing desert hike in the foothills of Borrego Springs. After an evening of food, drink, and lively exchange, I had the long-overdue epiphany to channel my training in musical performance more directly into my academic pursuits. The early support, friendship, and collegiality of these two had a tremendously positive effect on the arc of my scholarship; knowing they believed in the project helped me pencil its first sketchy contours—and ultimately see it through to the end.
    [Show full text]
  • Digital Signals
    Technical Information Digital Signals 1 1 bit t Part 1 Fundamentals Technical Information Part 1: Fundamentals Part 2: Self-operated Regulators Part 3: Control Valves Part 4: Communication Part 5: Building Automation Part 6: Process Automation Should you have any further questions or suggestions, please do not hesitate to contact us: SAMSON AG Phone (+49 69) 4 00 94 67 V74 / Schulung Telefax (+49 69) 4 00 97 16 Weismüllerstraße 3 E-Mail: [email protected] D-60314 Frankfurt Internet: http://www.samson.de Part 1 ⋅ L150EN Digital Signals Range of values and discretization . 5 Bits and bytes in hexadecimal notation. 7 Digital encoding of information. 8 Advantages of digital signal processing . 10 High interference immunity. 10 Short-time and permanent storage . 11 Flexible processing . 11 Various transmission options . 11 Transmission of digital signals . 12 Bit-parallel transmission. 12 Bit-serial transmission . 12 Appendix A1: Additional Literature. 14 99/12 ⋅ SAMSON AG CONTENTS 3 Fundamentals ⋅ Digital Signals V74/ DKE ⋅ SAMSON AG 4 Part 1 ⋅ L150EN Digital Signals In electronic signal and information processing and transmission, digital technology is increasingly being used because, in various applications, digi- tal signal transmission has many advantages over analog signal transmis- sion. Numerous and very successful applications of digital technology include the continuously growing number of PCs, the communication net- work ISDN as well as the increasing use of digital control stations (Direct Di- gital Control: DDC). Unlike analog technology which uses continuous signals, digital technology continuous or encodes the information into discrete signal states (Fig. 1). When only two discrete signals states are assigned per digital signal, these signals are termed binary si- gnals.
    [Show full text]
  • Reference Manual
    1 6 - V O I C E R E A L A N A L O G S Y N T H E S I Z E R REFERENCE MANUAL February 2001 Your shipping carton should contain the following items: 1. Andromeda A6 synthesizer 2. AC power cable 3. Warranty Registration card 4. Reference Manual 5. A list of the Preset and User Bank Mixes and Programs If anything is missing, please contact your dealer or Alesis immediately. NOTE: Completing and returning the Warranty Registration card is important. Alesis contact information: Alesis Studio Electronics, Inc. 1633 26th Street Santa Monica, CA 90404 USA Telephone: 800-5-ALESIS (800-525-3747) E-Mail: [email protected] Website: http://www.alesis.com Alesis Andromeda A6TM Reference Manual Revision 1.0 by Dave Bertovic © Copyright 2001, Alesis Studio Electronics, Inc. All rights reserved. Reproduction in whole or in part is prohibited. “A6”, “QCard” and “FreeLoader” are trademarks of Alesis Studio Electronics, Inc. A6 REFERENCE MANUAL 1 2 A6 REFERENCE MANUAL Contents CONTENTS Important Safety Instructions ....................................................................................7 Instructions to the User (FCC Notice) ............................................................................................11 CE Declaration of Conformity .........................................................................................................13 Introduction ................................................................................................................ 15 How to Use this Manual....................................................................................................................16
    [Show full text]
  • Chapter 1 Computers and Digital Basics Computer Concepts 2014 1 Your Assignment…
    Chapter 1 Computers and Digital Basics Computer Concepts 2014 1 Your assignment… Prepare an answer for your assigned question. Use Chapter 1 in the book and this PowerPoint to procure information. Prepare a PowerPoint presentation to: 1. Show your answer and additional information/facts – make sure you understand and can explain your answer. Provide as must information and detail as possible. Add graphics to enhance. 2. Where in the book did you find your information? Include the page number. 3. What more do you need to find out to help you better understand this question? Be prepared to share your information with the class. Chapter 1: Computers and Digital Basics 2 1 The Digital Revolution The digital revolution is an ongoing process of social, political, and economic change brought about by digital technology, such as computers and the Internet The technology driving the digital revolution is based on digital electronics and the idea that electrical signals can represent data, such as numbers, words, pictures, and music Chapter 1: Computers and Digital Basics 6 1 The Digital Revolution Digitization is the process of converting text, numbers, sound, photos, and video into data that can be processed by digital devices The digital revolution has evolved through four phases, beginning with big, expensive, standalone computers, and progressing to today’s digital world in which small, inexpensive digital devices are everywhere Chapter 1: Computers and Digital Basics 7 1 The Digital Revolution Chapter 1: Computers and Digital Basics
    [Show full text]
  • Improved Audio Filtering Using Extended High Pass Filters
    International Journal of Engineering Research & Technology (IJERT) ISSN: 2278-0181 Vol. 2 Issue 6, June - 2013 Improved Audio Filtering Using Extended High Pass Filters 1 2 Radhika Bhagat Ramandeep Kaur Department of Computer Science & Engineering, Guru Nanak Dev University Amritsar, Punjab, India Abstract—Noise Filtering in audio signals has always been a challenge since the noise is spread across a large bandwidth and overlaps the spectral range of the audio signal being recovered. In audio systems all these can be kept below the audible level but ambient noise may not be avoided even if the audio system is designed according to the best practices. So, there have been several audio filtering techniques such as spectral subtraction, Dolby noise reduction, use of low-pass and high pass filters, FIR and IIR filtering, etc. Noise filtering improvements were assessed for both noise reduction and signal degradation effects by different signal to noise ratio computations. Keywords: Audio Filtering, Low Pass Filters, High Pass Filters, FIR Filters, IIR Filters. I. INTRODUCTION With the development of communication technology, voice communication has become a major communication medium for people to transmit information more convenient. However, the widespread nature of noise makes the voice communication quality has declined. Therefore, to reduceIJERTIJERT the noise on the performance of voice communications, improve the quality of voice communications [1], voice denoising for technology has become a hot research topic. In this paper we will discuss various audio filtering techniques that will help in overcoming noise related issues. Audio Filter: - Is a frequency dependent amplifier circuit, working in the audio frequency range,0Hz- beyond 20 KHz.
    [Show full text]
  • SSB and the Phasing Exciter Hints and Kinks for Best Performance and Being Nice to One’S Neighbor’S
    SSB and the Phasing Exciter Hints and Kinks for Best Performance and Being Nice to One’s Neighbor’s Nick Tusa – K5EF How is SSB Generated? Brute Force v. Elegance A Tug or War in the 1950s One way is to design a steep-sided Steep-sided filters was expensive in the bandpass filter that passes one 1940-early 50s. Required carefully sideband and hugely attenuates the selected and ground individual crystals other. or expert machining of temperature- stable materials for a mechanical filter. Or Or We can use a mathematical analog of phase relationships to cancel the Audio filters having linear phase shift unwanted sideband and reinforce made with inexpensive Rs and Cs. the desired sideband. In Amateur Circles, Phasing Led the Way Don Norgaard really got the Amateur Ball rolling with his SSB, Jr. published in GE HAM News. (Refer to George W1LSB’s SSB discussion at last year’s W9DYV Event and CE website). Companies such as CE, Lakeshore Industries, Eldico, Johnson, Hallicrafters, Heathkit, B&W and others produced exciters using the Phasing Principal. By the late 1950s, filter technology improved and costs dropped, pushing Phasing aside…at least until the Software Defined Radio came along… Basic Phasing SSB Exciter Audio Phase Shift Linearity is Key to SB Suppression The AF Phase Shifter must maintain 90° differential across 300-3500Hz audio band. Small deviations result in degraded SB suppression i.e., a 2˚ difference = 35db suppression. Typical CE 10A/20A based on the Norgaard design = 40db if perfect! Central Electronics PS-1 Network Based on Norgaard’s SSB, Jr design.
    [Show full text]
  • Lecture Notes for Digital Electronics
    Lecture Notes for Digital Electronics Raymond E. Frey Physics Department University of Oregon Eugene, OR 97403, USA [email protected] March, 2000 1 Basic Digital Concepts By converting continuous analog signals into a finite number of discrete states, a process called digitization, then to the extent that the states are sufficiently well separated so that noise does create errors, the resulting digital signals allow the following (slightly idealized): • storage over arbitrary periods of time • flawless retrieval and reproduction of the stored information • flawless transmission of the information Some information is intrinsically digital, so it is natural to process and manipulate it using purely digital techniques. Examples are numbers and words. The drawback to digitization is that a single analog signal (e.g. a voltage which is a function of time, like a stereo signal) needs many discrete states, or bits, in order to give a satisfactory reproduction. For example, it requires a minimum of 10 bits to determine a voltage at any given time to an accuracy of ≈ 0:1%. For transmission, one now requires 10 lines instead of the one original analog line. The explosion in digital techniques and technology has been made possible by the incred- ible increase in the density of digital circuitry, its robust performance, its relatively low cost, and its speed. The requirement of using many bits in reproduction is no longer an issue: The more the better. This circuitry is based upon the transistor, which can be operated as a switch with two states. Hence, the digital information is intrinsically binary. So in practice, the terms digital and binary are used interchangeably.
    [Show full text]
  • Lecture 9 Analog and Digital I/Q Modulation
    Lecture 9 Analog and Digital I/Q Modulation Analog I/Q Modulation • Time Domain View •Polar View • Frequency Domain View Digital I/Q Modulation • Phase Shift Keying • Constellations 11/4/2006 Coherent Detection Transmitter Output 0 x(t) y(t) π 2cos(2 fot) Receiver Output Lowpass y(t) z(t) r(t) π 2cos(2 fot) • Requires receiver local oscillator to be accurately aligned in phase and frequency to carrier sine wave 11/4/2006 L Lecture 9 Fall 2006 2 Impact of Phase Misalignment in Receiver Local Oscillator Transmitter Output 0 x(t) y(t) π 2cos(2 fot) Receiver Output Output is zero Lowpass y(t) z(t) r(t) π 2sin(2 fot) • Worst case is when receiver LO and carrier frequency are phase shifted 90 degrees with respect to each other 11/4/2006 L Lecture 9 Fall 2006 3 Analog I/Q Modulation Baseband Input iti(t)(t) it (t) t cos 2 f tπ yt (t) 2cos(2( π 0 )f1t) π 2sin(2sin() 2π f0t f1t) qqt(t)(t) t qt (t) • Analog signals take on a continuous range of values (as viewed in the time domain) • I/Q signals are orthogonal and therefore can be transmitted simultaneously and fully recovered 11/4/2006 L Lecture 9 Fall 2006 4 Polar View of Analog I/Q Modulation it (t) = i(t)cos() 2π fot + 0° ii(t(t)t) it (t) qt (t) = q(t)cos() 2π fot + 90° = q(t)sin() 2π fot t cos 2π f tπ y (t) 2cos(2( 0 )f1t) t 2 2 π 2sin(2sin() 2π f0t f1t) yt (t) = i (t) + q (t) cos() 2π fot + θ(t) qq(tt(t)) −1 where θ(t) = tan q(t)/i(t) t qt (t) −180°<θ < 180° 11/4/2006 L Lecture 9 Fall 2006 5 Polar View of Analog I/Q Modulation (Con’t) • Polar View shows amplitude and phase of it(t), qt(t) and yt(t) combined signal for transmission at a given frequency f.
    [Show full text]
  • Introduction (Pdf)
    chapter1.fm Page 1 Thursday, August 17, 2000 4:43 PM CHAPTER 1 INTRODUCTION The evolution of digital circuit design n Compelling issues in digital circuit design n How to measure the quality of digital design n Valuable references 1.1 A Historical Perspective 1.2 Issues in Digital Integrated Circuit Design 1.3 Quality Metrics of A Digital Design 1.4 Summary 1.5 To Probe Further 1 chapter1.fm Page 2 Thursday, August 17, 2000 4:43 PM 2 INTRODUCTION Chapter 1 1.1A Historical Perspective The concept of digital data manipulation has made a dramatic impact on our society. One has long grown accustomed to the idea of digital computers. Evolving steadily from main- frame and minicomputers, personal and laptop computers have proliferated into daily life. More significant, however, is a continuous trend towards digital solutions in all other areas of electronics. Instrumentation was one of the first noncomputing domains where the potential benefits of digital data manipulation over analog processing were recognized. Other areas such as control were soon to follow. Only recently have we witnessed the con- version of telecommunications and consumer electronics towards the digital format. Increasingly, telephone data is transmitted and processed digitally over both wired and wireless networks. The compact disk has revolutionized the audio world, and digital video is following in its footsteps. The idea of implementing computational engines using an encoded data format is by no means an idea of our times. In the early nineteenth century, Babbage envisioned large- scale mechanical computing devices, called Difference Engines [Swade93]. Although these engines use the decimal number system rather than the binary representation now common in modern electronics, the underlying concepts are very similar.
    [Show full text]
  • A Micro-Programmable Correlator for Real-Time Radar Processing
    ALKER: Correlator for radar processing A MICRO-PROGRAMMABLE CORRELATOR FOR REAL-TIME RADAR PROCESSING by Hans-J¢rgen Alker ELECTRONICS RESEARCH LABORATORY (ELAB) Trondheim, NORWAY ABSTRACT This paper presents a novel design of a mul tibi t ·, digital correlator for incoherent scatter radar observations. By utilizing bit-slice microprocessor elements and internal control by microcode instructions, a high-speed arithmetical preprocessor has been developed. Arithmetical and control operations are separated to obtain increased speed performance. The arithmetical part is optimized for complex auto/cross correlation processing and has an effective rate of 24·106 multiplications/sec. Input data has 8-bit accuracy in integer format. The processor includes, at input, a 2-ported buffer memory for storing single/multiple channel inputs. Processing results are temporarily stored in a 4 K word high-speed memory before transferred to a general-purpose computer. Processing speed can be increased by adding up to 3 slave processors thus achieving direct parallel processing. Internal hardware is available for interface with standard CAMAC modules. INTRODUCTION During 1976-79 a feasibility study of a digital, multibit correlator for the EISCAT (European Incoherent SCATter) radar system was conducted. The main topics of the study embraced system design and hardware construction of a digital processor which fulfilled the specifications for real-time data process- ing. The result of the study is a prototype construction of a high- speed multiprocessor system under interactive control of the EISCAT radar site computer. Support software for microprogram development, simulation and hardware testing are available for execution on a host computer. The main objectives for constructing a digital processor for the EISCAT system were the ultimate requirements for special purpose data handling algorithms and the real-time processing speed.
    [Show full text]
  • Hardware Design Techniques
    HARDWARE DESIGN TECHNIQUES ANALOG-DIGITAL CONVERSION 1. Data Converter History 2. Fundamentals of Sampled Data Systems 3. Data Converter Architectures 4. Data Converter Process Technology 5. Testing Data Converters 6. Interfacing to Data Converters 7. Data Converter Support Circuits 8. Data Converter Applications 9. Hardware Design Techniques 9.1 Passive Components 9.2 PC Board Design Issues 9.3 Analog Power Supply Systems 9.4 Overvoltage Protection 9.5 Thermal Management 9.6 EMI/RFI Considerations 9.7 Low Voltage Logic Interfacing 9.8 Breadboarding and Prototyping I. Index ANALOG-DIGITAL CONVERSION HARDWARE DESIGN TECHNIQUES 9.1 PASSIVE COMPONENTS CHAPTER 9 HARDWARE DESIGN TECHNIQUES This chapter, one of the longer of those within the book, deals with topics just as important as all of those basic circuits immediately surrounding the data converter, discussed earlier. The chapter deals with various and sundry circuit/system issues which fall under the guise of system hardware design techniques. In this context, the design techniques may be all those support items surrounding a data converter, excluding the data converter itself. This includes issues of passive components, printed circuit design, power supply systems, protection of linear devices against overvoltage and thermal effects, EMI/RFI issues, high speed logic considerations, and finally, simulation, breadboarding and prototyping. Some of these topics aren't directly involved in the actual signal path of a design, but they are every bit as important as choosing the correct device and surrounding circuit values. Remote sensing and signal conditioning is such a vital part of data conversion that a considerable amount of discussion is given to topics such as overvoltage protection, cable driving, shielding, and receiving—where the remote interface is often with op amps and instrumentation amplifiers.
    [Show full text]
  • MAS.632 Conversational Computer Systems
    MIT OpenCourseWare http://ocw.mit.edu MAS.632 Conversational Computer Systems Fall 2008 For information about citing these materials or our Terms of Use, visit: http://ocw.mit.edu/terms. 10 Basics of Telephones This and the following chapters explain the technology and computer applica­ tions oftelephones, much as earlier pairs ofchapters explored speech coding, syn­ thesis, and recognition. The juxtaposition of telephony with voice processing in a single volume is unusual; what is their relationship? First, the telephone is an ideal means to access interactive voice response services such as those described in Chapter 6. Second, computer-based voice mail will give a strong boost to other uses of stored voice such as those already described in Chapter 4 and to be explored again in Chapter 12. Finally, focusing on communication,i.e., the task rather than the technology, leads to a better appreciation of the broad intersec­ tion of speech, computers, and our everyday work lives. This chapter describes the basic telephone operations that transport voice across a network to a remote location. Telephony is changing rapidly in ways that radically modify how we think about and use telephones now and in the future. Conventional telephones are already ubiquitous for the business traveler in industrialized countries, but the rise in personal, portable, wireless telephones is spawning entirely new ways of thinking about universal voice connectivity. The pervasiveness of telephone and computer technologies combined with the critical need to communicate in our professional lives suggest that it would be foolish to ignore the role of the telephone as a speech processing peripheral just like a speaker or microphone.
    [Show full text]