DOCSLIB.ORG

Selection and Operation of AUDIO SIGNAL Processors

Total Page：16

File Type：pdf, Size：1020Kb

Download full-text PDF Read full-text

Load more

SELECTION AND OPERATION AUDIO SIGNAL PROCESSORS By Gino Sigismondi A Shure Educational Publication Selection and Operation of Table of Contents AUDIO SIGNAL Processors Introduction . 4 What Are Audio Signal Processors? . 4 What Types of Problems Can Benefit from Audio Processing? . 5 Feedback . 6 Chapter 1 Types of Audio Processors . 7 1.1 Volume (Gain) Control . 7 1.2 Filters and Equalization . 7 1.3 Dynamics Processors . 12 1.4 Delay . 16 1.5 Adaptive Audio Processors . 17 Chapter 2 Practical Applications For Audio Signal Processors . 23 2.1 Maximizing Gain-Before-Feedback . 23 2.2 Improving Speech Intelligibility . 24 2.3 Sound System Gain Structure . 26 2.4 Digital Signal Processing . 28 Reference Information . 30 Appendix A: Sound Waves . 30 Appendix B: Potential Acoustic Gain (PAG) and Needed Acoustic Gain (NAG) . 32 Glossary . 34 Shure Product Selection Charts . 36 Bibliography . 38 Acknowledgements . 38 About the Author . 39 Audio Signal Processors 3 Selection and Operation of AUDIO SIGNAL Processors Introduction For any sound system, the primary goal is good if the three primary measures are not satisfied, any sound. What, however, constitutes "good" sound? The subjective terms take on even less importance. Speech three primary measures of good sound are audibility, that is "warm" but unintelligible does the listener little good. intelligibility, and fidelity. Many factors contribute to the Audio signal processors offer a variety of tools to assist quality of the sound, including the quality of the sound in optimizing a sound system for audibility, intelligibility, sources, the sound system, and the room acoustics. and fidelity. While not usually essential for a sound system The audibility of speech or music at the furthest to operate (i.e., provide high-level sound reinforcement listener must be sufficient to achieve the desired effect: of low-level sources), audio signal processors can be usually a comfortable listening level for speech, and more invaluable tools in sound system design. A basic sound powerful levels for certain kinds of music. These levels system consists of four components: should be attainable without distortion or feedback. • Input devices (microphones, CD players, etc) Intelligibility is determined by the signal-to-noise ratio and • Mixers (to combine inputs, control levels, direct-to- reverberant ratio at the listener’s ear. The "signal" and provide preamplification, if necessary) is the desired sound source (speech, musical instruments, • Amplifiers etc.), while the "noise" is ambient sound in the room as • Output devices (loudspeakers) well as electrical noise produced by the sound system. Audio signal processors are typically employed Maximum speech intelligibility requires a speech level of within or just after the mixer stage, but before amplification. at least 20 dB above the noise floor at the listener’s ear. (See Figure 1-1.) A processor can be used at the input The direct-to-reverberant ratio is determined by the stage, but since most processors are designed to operate directivity of the loudspeakers and the reverberation with line level sources this is rare. Signal processors can characteristics of the room. High levels of reverberation be analog or digital, single- or multi-function, stand-alone can severely degrade intelligibility by making it difficult to devices or integrated with other components in the sound distinguish the end of one word and the start of the next. system. Most signal processors originated as stand alone Finally, fidelity of sound is primarily defined by the overall devices designed for a specific purpose. Over time, frequency response of the sound arriving at the listener’s integration of similar processors into one device became ear. The frequency range must be sufficiently wide and popular (e.g. compressor/limiters). The development of relatively uniform in order to provide realistic and accurate audio processors that operate in the digital domain allowed reinforcement of speech and music. Every component in for further integration, leading to multi-function digital signal the signal chain contributes to this, and a limitation at any processors (DSP) that combine seemingly disparate point will affect the fidelity of the entire system. functions into a single unit. Perhaps more importantly, DSP Other more subjective terms may be applied to good devices offer these functions at a cost that is a fraction of sound ("warmth", "punch", etc.), but these colloquialisms the purchase price of several individual processors. are not measurable in any meaningful way. Additionally, Introduction 4 Selection and Operation of AUDIO SIGNAL Processors INSTRUMENT MICROPHONE AMPLIFIER LOUDSPEAKER VOCAL MICROPHONE BOUNDARY MIXER PROCESSOR MICROPHONE Figure 1-1: basic sound system What Types of Problems Can Benefit from effect on the sound. Reverberation can be reduced only by Audio Processing? absorptive acoustic treatment or structural modification; To understand the purpose of audio signal processing, electronics cannot remove it. If additional acoustic it is necessary to examine the problems encountered in treatment is not an option, directional loudspeakers allow a typical sound system. Note that an audio processor the sound to be "aimed" toward the listener and away from cannot solve all the potential problems in a sound reflective surfaces. Simply raising the level of the sound reinforcement system. The most common problems are system will only aggravate the problem by raising the listed to the on the next page: reverberation level as well. Long reverberation times The importance of good room acoustics cannot be severely reduce intelligibility. In audio teleconferencing underestimated. In any room where sound reinforcement systems, this results in a hollow, or "bottom-of-the-barrel" will be used, excess reverberation times introduce a sound received by the remote site. myriad of problems that cannot be solved by any audio processors. Reverberation time is the length of time that a sound persists in a room after the sound source has Want to know more about proper microphone usage? stopped. All attempts should be made to keep unwanted Shure offers the following educational guides free sounds from entering the microphone in the first place. of charge: The level of desired sound at the microphone should be • Microphone Techniques for Studio Recording at least 30 dB above any ambient sound picked up by the microphone. Proper microphone placement (a full • Microphone Techniques for Live Sound discussion of which is beyond the scope of this publication) Reinforcement is also crucial. A good rule of thumb: always keep Visit shure.com or contact your local Shure microphones as close as possible to the sound source. office (see back cover) to request your Once sound energy is introduced into the acoustic complimentary copies. space by the loudspeaker, processing no longer has any 5 Selection and Operation of AUDIO SIGNAL Processors Feedback Problems: Remedies: Feedback is characterized by a sustained, Feedback Parametric Equalizer/ ringing tone, which can vary from a low rumble to a Automatic Mixer/ piercing screech. Echoes and reverberation caused Feedback Reducer by room acoustics, as well as ground buzz and other Poor tone quality (subjective) Graphic equalizer extraneous noises, are not the same thing as feed- Sound source too loud Compressor/Limiter/AGC back, and cannot be cured in the same manner. Sound source too quiet AGC Feedback occurs whenever the sound entering a microphone is reproduced by a loudspeaker, picked Varying signal levels Compressor/Limiter/AGC up by the microphone, and re-amplified again and from multiple sound sources again. The familiar howl of feedback is an oscillation Unwanted noise Noisegate/Downward expander that is triggered by sound entering the microphone. Unexpected transients Compressor/Limiter/No overshot The easiest way to (intentionally) create feedback is to ("Look-ahead") Peak Limiter point a microphone directly into a loudspeaker. Comb filtering Automatic Microphone Mixer Placing the microphone too close to the loudspeaker, too far from the sound source, or simply turning the due to open microphones microphone up too loud exacerbates feedback Frequency response Delay problems. Other contributing factors are too many anomalies due to open microphones, poor room acoustics, and uneven misaligned loudspeakers frequency response in either the microphones or Poor intelligibility Parametric Equalizer/ loudspeakers. Automatic Microphone Mixer The single easiest way to reduce feedback is to move the microphone closer to the desired sound Acoustic echoes Acoustic Echo Canceller source. Additionally, using a directional microphone (in teleconferencing systems) (cardioid, supercardioid, etc.) will slightly increase the Distortion Compressor/Limiter amount of gain-before-feedback. Reducing the (due to wide dynamic range) number of open microphones with an automatic mixer will also improve the situation. Try to keep Problems that cannot be solved by audio processing: microphones and lou speakers as far away from each other as possible. Lastly, acoustically treat the room • Echoes because of poor room acoustics to cover hard, reflective surfaces such as glass, • Poor sound due to excess room reverberation times marble, and wood. Realize, though, that in certain • Feedback caused by operating beyond the limits of PAG rooms long reverberation times may be desirable, (see Appendix 2) such as a house of worship used for acoustic music performance. • Noise
Recommended publications
  • AUTOMATIC MIXERS Written by Tom Stuckman and Steve Marks

    AUTOMATIC MIXERS Written by Tom Stuckman and Steve Marks

    1 AUTOMATIC MIXERS Written by Tom Stuckman and Steve Marks The use of sound reinforcement systems is common place in today’s churches, boardrooms, meeting rooms, auditoriums and other facilities. In fact, it is hard to imagine what it was like before their wide spread use. With advances in the sound quality in the entertainment industry, television, CDs and home theatre for example, the expectation of quality sound reinforcement has risen as well. When sound system requirements progress past the need of a single microphone, then mixing becomes an issue. Depending on how the system is operated, the performance of a sound system with multiple microphones can also change for the better, or worse. Automatic mixers are an important, and often misunderstood, tool of the sound system designer that can improve gain before feedback, improve intelligibility and reduce inter-microphone interference (comb filtering) in many multi-microphone systems. To begin with, let us briefly examine why we use a sound reinforcement system. A sound reinforcement system is generally used to aid the listener by increasing the volume of a sound source sufficiently above the background noise so that it can be easily heard. Another benefit, when used in an acoustic environment, is to improve the clarity of the sound source at the listener. Stated another way, the goal is to make the sound source (talker) appear to be closer to the listener. There are several factors that influence how a sound system meets these goals, but we will concentrate on the microphone and microphone mixer and their use in an acoustic space.
  • And Passive Speakers?

    And Passive Speakers?

    To be active or not to be active – that is the question... To be active or not to be active – that is the question... 1. Active, passive – the situation 2. Active and passive loudspeaker – the basic difference 3. Passive loudspeaker 4. Active loudspeaker 5. The ADAM loudspeaker: passive option, active optimum 1. Active versus Passive – the Situation In any hifi-system, the loudspeakers are the pivotal component concerning sound quality. That is not to say that the other components do not matter. Nevertheless, it is indisputable that the loudspeaker is decisive for the sound of a hifi-system. It is – besides the acoustical properties of the listening room and the recording itself – the core of any music reproduction. The history of loudspeaker development has produced a great variety of very different systems and designs. The circuit technology of the frequency-separating filter that separates the audio signal into different frequency ranges is determining the design of a loudspeaker. In this respect we distinguish between active and passive systems. Usually, this is a topic that is often underestimated in its importance for sound quality. Active-passive is much more than just a technical negligibility: In fact, the impact of the dividing network on the overall sound of a loudspeaker is substantial. Active or passive – which system is preferable? Considering the aspects mentioned before, it may become a little more comprehensive why the very question comes up over and over again in the hifi-world. For decades it has been spooking as a debate on principles in the journals and magazines and for some time, now, in the web forums.
  • Real-Time Programming and Processing of Music Signals Arshia Cont

    Real-Time Programming and Processing of Music Signals Arshia Cont

    Real-time Programming and Processing of Music Signals Arshia Cont To cite this version: Arshia Cont. Real-time Programming and Processing of Music Signals. Sound [cs.SD]. Université Pierre et Marie Curie - Paris VI, 2013. tel-00829771 HAL Id: tel-00829771 https://tel.archives-ouvertes.fr/tel-00829771 Submitted on 3 Jun 2013 HAL is a multi-disciplinary open access L’archive ouverte pluridisciplinaire HAL, est archive for the deposit and dissemination of sci- destinée au dépôt et à la diffusion de documents entific research documents, whether they are pub- scientifiques de niveau recherche, publiés ou non, lished or not. The documents may come from émanant des établissements d’enseignement et de teaching and research institutions in France or recherche français ou étrangers, des laboratoires abroad, or from public or private research centers. publics ou privés. Realtime Programming & Processing of Music Signals by ARSHIA CONT Ircam-CNRS-UPMC Mixed Research Unit MuTant Team-Project (INRIA) Musical Representations Team, Ircam-Centre Pompidou 1 Place Igor Stravinsky, 75004 Paris, France. Habilitation à diriger la recherche Defended on May 30th in front of the jury composed of: Gérard Berry Collège de France Professor Roger Dannanberg Carnegie Mellon University Professor Carlos Agon UPMC - Ircam Professor François Pachet Sony CSL Senior Researcher Miller Puckette UCSD Professor Marco Stroppa Composer ii à Marie le sel de ma vie iv CONTENTS 1. Introduction1 1.1. Synthetic Summary .................. 1 1.2. Publication List 2007-2012 ................ 3 1.3. Research Advising Summary ............... 5 2. Realtime Machine Listening7 2.1. Automatic Transcription................. 7 2.2. Automatic Alignment .................. 10 2.2.1.
  • Introduction to the Digital Snake

    Introduction to the Digital Snake

    TABLE OF CONTENTS What’s an Audio Snake ........................................4 The Benefits of the Digital Snake .........................5 Digital Snake Components ..................................6 Improved Intelligibility ...........................................8 Immunity from Hums & Buzzes .............................9 Lightweight & Portable .......................................10 Low Installation Cost ...........................................11 Additional Benefits ..............................................12 Digital Snake Comparison Chart .......................14 Conclusion ...........................................................15 All rights reserved. No part of this publication may be reproduced in any form without the written permission of Roland System Solutions. All trade- marks are the property of their respective owners. Roland System Solutions © 2005 Introduction Digital is the technology of our world today. It’s all around us in the form of CDs, DVDs, MP3 players, digital cameras, and computers. Digital offers great benefits to all of us, and makes our lives easier and better. Such benefits would have been impossible using analog technology. Who would go back to the world of cassette tapes, for example, after experiencing the ease of access and clean sound quality of a CD? Until recently, analog sound systems have been the standard for sound reinforcement and PA applications. However, recent technological advances have brought the benefits of digital audio to the live sound arena. Digital audio is superior
  • Audio Interface User's Manual

    Audio Interface User's Manual

    Audio Interface User’s Manual Eurorack Synthesizer Modules 14 HP TABLE OF CONTENTS 1.INTRODUCTION 2.WARRANTY 3.INSTALLATION 4.FUNCTION OF PANEL COMPONENTS 5.SIGNALFLOW & ROUTING 6.SPECIFICATIONS 2 1. INTRODUCTION Audiophile Circuits League. -The main purpose of the ACL Audio Interface module is to interface modular synthesizer systems with professional audio recording and stage equipment. The combination of studio quality signal path, �lexible routing possibilities and a headphoneheadphones ampli�ier, with low capabledistortion, of drivingmakes theboth connection high and low between impedance these different environments effortless and sonically transparent. The ACL Audio Interface offers balanced to unbalanced and unbalanced to balanced stereo lines with level controls. The stereo signal from the auxiliary input,either alsowith with balanced level tocontrol, unbalanced, can be oroptionally unbalanced routed to balanced to and mixed line signals,together or can be muted. The headphone ampli�ier can also get its signal from one or the other line after the level control and mixing stage, or can be muted. Since the ampli�ier is AC coupled only at the input, but not at the output, there is an on-board DC protection circuit included. In case the headphone ampli�ier is driven into clipping, the protection can also be tripped. The module has a soft start function* and one overload indicator for every line. *With the soft start function, the interface switch is turned on after a while after turning on the Eurorack main unit. This function can prevent output of unexpecteddamage to the sound speaker. that another module will emit at startup, which will cause 3 2.
  • 21065L Audio Tutorial

    21065L Audio Tutorial

    a Using The Low-Cost, High Performance ADSP-21065L Digital Signal Processor For Digital Audio Applications Revision 1.0 - 12/4/98 dB +12 0 -12 Left Right Left EQ Right EQ Pan L R L R L R L R L R L R L R L R 1 2 3 4 5 6 7 8 Mic High Line L R Mid Play Back Bass CNTR 0 0 3 4 Input Gain P F R Master Vol. 1 2 3 4 5 6 7 8 Authors: John Tomarakos Dan Ledger Analog Devices DSP Applications 1 Using The Low Cost, High Performance ADSP-21065L Digital Signal Processor For Digital Audio Applications Dan Ledger and John Tomarakos DSP Applications Group, Analog Devices, Norwood, MA 02062, USA This document examines desirable DSP features to consider for implementation of real time audio applications, and also offers programming techniques to create DSP algorithms found in today's professional and consumer audio equipment. Part One will begin with a discussion of important audio processor-specific characteristics such as speed, cost, data word length, floating-point vs. fixed-point arithmetic, double-precision vs. single-precision data, I/O capabilities, and dynamic range/SNR capabilities. Comparisions between DSP's and audio decoders that are targeted for consumer/professional audio applications will be shown. Part Two will cover example algorithmic building blocks that can be used to implement many DSP audio algorithms using the ADSP-21065L including: Basic audio signal manipulation, filtering/digital parametric equalization, digital audio effects and sound synthesis techniques. TABLE OF CONTENTS 0. INTRODUCTION ................................................................................................................................................................4 1.
  • Ds101 Noise Gate for the 500 Series Rack System

    Ds101 Noise Gate for the 500 Series Rack System

    DRAWMER DS101 NOISE GATE FOR THE 500 SERIES RACK SYSTEM CONTENTS Warranty . 2 Safety Consideration . 2 Radio Frequencies Statement . 2 Chapter 1 - Introduction Introduction . 3 Installation . 4 Audio Connection . 4 Typical Connection Guide . 5 Chapter 2 - Control Description The Gate Controls . 6 The Key Filters . 8 Quick Start Guide . 9 Noise Gate Operation . .10 Chapter 3 - General Information If a fault develops. 15 Contacting Drawmer . .15 Specification. 15 Block Diagram. 16 Session Recall Sheet . 17 DRAWMER COPYRIGHT This manual is copyrighted © 2012 by Drawmer Electronics Ltd. With all rights reserved. Under copyright laws, no part of this publication may be reproduced, transmitted, stored in a retrieval system or translated into any language in any form by any means, mechanical, optical, electronic, recording, or otherwise, without the written permission of Drawmer Electronics Ltd. ONE YEAR LIMITED WARRANTY Drawmer Electronics Ltd., warrants the Drawmer DS101 Noise Gate For the USA to conform substantially to the specifications of this manual for a period of one year from the original date of purchase when used in FEDERAL COMMUNICATIONS COMMISSION RADIO accordance with the specifications detailed in this manual. In the FREQUENCY INTERFERENCE STATEMENT case of a valid warranty claim, your sole and exclusive remedy and This equipment has been tested and found to comply with the limits Drawmer’s entire liability under any theory of liability will be to, at for a Class B digital device, pursuant to Part 15 of the FCC Rules. Drawmer’s discretion, repair or replace the product without charge, These limits are designed to provide reasonable protection against or, if not possible, to refund the purchase price to you.
  • User's Manual

    User's Manual

    USER’S MANUAL G-Force GUITAR EFFECTS PROCESSOR IMPORTANT SAFETY INSTRUCTIONS The lightning flash with an arrowhead symbol The exclamation point within an equilateral triangle within an equilateral triangle, is intended to alert is intended to alert the user to the presence of the user to the presence of uninsulated "dan- important operating and maintenance (servicing) gerous voltage" within the product's enclosure that may instructions in the literature accompanying the product. be of sufficient magnitude to constitute a risk of electric shock to persons. 1 Read these instructions. Warning! 2 Keep these instructions. • To reduce the risk of fire or electrical shock, do not 3 Heed all warnings. expose this equipment to dripping or splashing and 4 Follow all instructions. ensure that no objects filled with liquids, such as vases, 5 Do not use this apparatus near water. are placed on the equipment. 6 Clean only with dry cloth. • This apparatus must be earthed. 7 Do not block any ventilation openings. Install in • Use a three wire grounding type line cord like the one accordance with the manufacturer's instructions. supplied with the product. 8 Do not install near any heat sources such • Be advised that different operating voltages require the as radiators, heat registers, stoves, or other use of different types of line cord and attachment plugs. apparatus (including amplifiers) that produce heat. • Check the voltage in your area and use the 9 Do not defeat the safety purpose of the polarized correct type. See table below: or grounding-type plug. A polarized plug has two blades with one wider than the other.
  • Low Power Asynchronous Digital Signal Processing

    Low Power Asynchronous Digital Signal Processing

    LOW POWER ASYNCHRONOUS DIGITAL SIGNAL PROCESSING A thesis submitted to the University of Manchester for the degree of Doctor of Philosophy in the Faculty of Science & Engineering October 2000 Michael John George Lewis Department of Computer Science 1 Contents Chapter 1: Introduction ....................................................................................14 Digital Signal Processing ...............................................................................15 Evolution of digital signal processors ....................................................17 Architectural features of modern DSPs .........................................................19 High performance multiplier circuits .....................................................20 Memory architecture ..............................................................................21 Data address generation .........................................................................21 Loop management ..................................................................................23 Numerical precision, overflows and rounding .......................................24 Architecture of the GSM Mobile Phone System ...........................................25 Channel equalization ..............................................................................28 Error correction and Viterbi decoding ...................................................29 Speech transcoding ................................................................................31 Half-rate and enhanced
  • Using Equalization on HF SSB

    Using Equalization on HF SSB

    Using Equalization on HF SSB Bill Leonard NØCU 4/27/2010 1 Topics: •Some Commonly Used Methods for Improving HF SSB Comms •Some key points about speech and hearing •The W2IHY 8 Band Equalizer + Noise Gate •What is it •When to use it •Where to use it •How to use it •What it can, and cannot do 4/27/2010 2 What is “communication”? •Communication Transfer of Information •What is “information transfer”? •CW •Digital comms •SSB Voice Different modes require different •Rag-chew methods to optimize information transfer •Breaking DX pile-ups •FM Voice 4/27/2010 3 What can we do to improve our ability to communicate via HF SSB? Typical communication path Processor Voice Mic EQ Xmtr Rcvr EQ Spkr Ear (Brain) .. .. Typical Why Not Here? Location 4/27/2010 4 Some Commonly Used Methods for Improving HF SSB Comms: 1. Improve received SNR: •Use higher gain antennas •Use higher peak transmitter power •Raise average transmit power (compression) •There is a limit: trade-off between distortion vs. SNR improvement •Some (W2IHY) claim that straight compression can degrade transmit SNR •I question this claim (all limiters exhibit “small signal suppression”) •Compression will increase background noise when no speech signal is present •Use of a Noise Gate should mitigate this problem •“Matched Filter” detection: •“Matching” filters means more than just reducing bandwidth arbitrarily •There is a limit: trade-off between distortion vs. SNR improvement •A 10 Hz filter won’t work very well with a 60 wpm CW signal •Reducing receiver noise figure will not help when atmospheric noise is dominant 4/27/2010 5 Some Commonly Used Methods for Improving HF SSB Comms (continued): 1.
  • The Digital Signal Processor Derby

    The Digital Signal Processor Derby

    SEMICONDUCTORS The newest breeds trade off speed, energy consumption, and cost to vie for The an ever bigger piece of the action Digital Signal Processor BY JENNIFER EYRE Derby Berkeley Design Technology Inc. pplications that use digital signal-processing purpose processors typically lack these specialized features and chips are flourishing, buoyed by increasing per- are not as efficient at executing DSP algorithms. formance and falling prices. Concurrently, the For any processor, the faster its clock rate or the greater the market has expanded enormously, to an esti- amount of work performed in each clock cycle, the faster it can mated US $6 billion in 2000. Vendors abound. complete DSP tasks. Higher levels of parallelism, meaning the AMany newcomers have entered the market, while established ability to perform multiple operations at the same time, have companies compete for market share by creating ever more a direct effect on a processor’s speed, assuming that its clock novel, efficient, and higher-performing architectures. The rate does not decrease commensurately. The combination of range of digital signal-processing (DSP) architectures available more parallelism and faster clock speeds has increased the is unprecedented. speed of DSP processors since their commercial introduction In addition to expanding competition among DSP proces- in the early 1980s. A high-end DSP processor available in sor vendors, a new threat is coming from general-purpose 2000 from Texas Instruments Inc., Dallas, for example, is processors with DSP enhancements. So, DSP vendors have roughly 250 times as fast as the fastest processor the company begun to adapt their architectures to stave off the outsiders.
  • Analog Signal Processing Christophe Caloz, Fellow, IEEE, Shulabh Gupta, Member, IEEE, Qingfeng Zhang, Member, IEEE, and Babak Nikfal, Student Member, IEEE

    Analog Signal Processing Christophe Caloz, Fellow, IEEE, Shulabh Gupta, Member, IEEE, Qingfeng Zhang, Member, IEEE, and Babak Nikfal, Student Member, IEEE

    1 Analog Signal Processing Christophe Caloz, Fellow, IEEE, Shulabh Gupta, Member, IEEE, Qingfeng Zhang, Member, IEEE, and Babak Nikfal, Student Member, IEEE Abstract—Analog signal processing (ASP) is presented as a discrimination effects, emphasizing the concept of group delay systematic approach to address future challenges in high speed engineering, and presenting the fundamental concept of real- and high frequency microwave applications. The general concept time Fourier transformation. Next, it addresses the topic of of ASP is explained with the help of examples emphasizing basic ASP effects, such as time spreading and compression, the “phaser”, which is the core of an ASP system; it reviews chirping and frequency discrimination. Phasers, which repre- phaser technologies, explains the characteristics of the most sent the core of ASP systems, are explained to be elements promising microwave phasers, and proposes corresponding exhibiting a frequency-dependent group delay response, and enhancement techniques for higher ASP performance. Based hence a nonlinear phase response versus frequency, and various on ASP requirements, found to be phaser resolution, absolute phaser technologies are discussed and compared. Real-time Fourier transformation (RTFT) is derived as one of the most bandwidth and magnitude balance, it then describes novel fundamental ASP operations. Upon this basis, the specifications synthesis techniques for the design of all-pass transmission of a phaser – resolution, absolute bandwidth and magnitude and reflection phasers