Author: Mitch Barnett
Publisher:
ISBN: 9781520977904
Category :
Languages : en
Pages : 204
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Book Description
This eBook provides the audio enthusiast with an easy-to-follow step-by-step guide for designing a custom digital filter that corrects the frequency and timing response of your loudspeakers in your listening environment so that the music arriving at your ears matches as closely as possible to the content on the recording.Industry guidelines, spanning over 40 years of evolution, are referenced throughout the eBook, providing the recommended target responses for accurate sound reproduction. Correcting the measured response to known target responses is, in effect, matching the acoustic output signal arriving at your ears to the digital audio stored in a media file. The target responses can also be tailored to your own preferences.The chapter on the acoustics and psychoacoustics of room correction explains why we hear what we hear in small-room acoustics and what our ears care about most when it comes to accurate sound reproduction. There are individual chapters on industry target responses for frequency and step response, group delay, energy time curve, reverb time, polar response and interaural coherence coefficient (i.e., imaging).Advanced sections of the eBook show you how to design and implement digital crossovers, driver linearization, and driver time alignment; how to compensate for brickwall antialiasing filters; and how to use the Beamforming quasi-anechoic measurement technique. All of these sections are designed to incrementally improve your sound reproduction system's accuracy. The last section of the eBook shows you how to perform loopback measurements to verify the design and operation of your digital correction filter, not only at the listening position, but across a wide listening area as well.Whether your sound reproduction system is a computer desktop speaker system, stereo audiophile system, home theater multi-channel system, mains and sub-woofer system, DIY audio system, or car sound system, the results are audible, measurable, and repeatable.To benefit the most from this step-by-step guide, you will require a computer, a calibrated measurement microphone, an Analog to Digital (ADC)/Digital to Analog (DAC) converter, and two commercial software packages: Acourate DSP Audio Toolbox and JRiver Media Center. This eBook contains several color-coded images, so viewing on a color screen is required.With over 260 color screen captures of the software in action, plus 200 links to industry reference material and knowledge, this eBook is the definitive guide to Accurate Sound Reproduction Using Digital Signal Processing (DSP).
Author: Floyd E. Toole
Publisher: Routledge
ISBN: 1317415094
Category : Technology & Engineering
Languages : en
Pages : 956
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Book Description
Sound Reproduction: The Acoustics and Psychoacoustics of Loudspeakers and Rooms, Third Edition explains the physical and perceptual processes that are involved in sound reproduction and demonstrates how to use the processes to create high-quality listening experiences in stereo and multichannel formats. Understanding the principles of sound production is necessary to achieve the goals of sound reproduction in spaces ranging from recording control rooms and home listening rooms to large cinemas. This revision brings new science-based perspectives on the performance of loudspeakers, room acoustics, measurements and equalization, all of which need to be appropriately used to ensure the accurate delivery of music and movie sound tracks from creators to listeners. The robust website (www.routledge.com/cw/toole) is the perfect companion to this necessary resource.
Author: John Watkinson
Publisher: CRC Press
ISBN: 1136118543
Category : Technology & Engineering
Languages : en
Pages : 576
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Book Description
Designed to make life a little easier by providing all the theoretical background necessary to understand sound reproduction, backed up with practical examples. Specialist terms - both musical and physical - are defined as they occur and plain English is used throughout. Analog and digital audio are considered as alternatives, and the advantages of both are stressed. Audio is only as good as the transducers employed, and consequently microphone and loudspeaker technology also feature heavily - making this the most comprehensive, up-to-date text currently available on all aspects of sound reproduction.
Author: John W. Leis
Publisher: John Wiley & Sons
ISBN: 1118033809
Category : Science
Languages : en
Pages : 352
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Book Description
Quickly Engages in Applying Algorithmic Techniques to Solve Practical Signal Processing Problems With its active, hands-on learning approach, this text enables readers to master the underlying principles of digital signal processing and its many applications in industries such as digital television, mobile and broadband communications, and medical/scientific devices. Carefully developed MATLAB® examples throughout the text illustrate the mathematical concepts and use of digital signal processing algorithms. Readers will develop a deeper understanding of how to apply the algorithms by manipulating the codes in the examples to see their effect. Moreover, plenty of exercises help to put knowledge into practice solving real-world signal processing challenges. Following an introductory chapter, the text explores: Sampled signals and digital processing Random signals Representing signals and systems Temporal and spatial signal processing Frequency analysis of signals Discrete-time filters and recursive filters Each chapter begins with chapter objectives and an introduction. A summary at the end of each chapter ensures that one has mastered all the key concepts and techniques before progressing in the text. Lastly, appendices listing selected web resources, research papers, and related textbooks enable the investigation of individual topics in greater depth. Upon completion of this text, readers will understand how to apply key algorithmic techniques to address practical signal processing problems as well as develop their own signal processing algorithms. Moreover, the text provides a solid foundation for evaluating and applying new digital processing signal techniques as they are developed.
Author: Franz Zotter
Publisher: Springer
ISBN: 3030172074
Category : Technology & Engineering
Languages : en
Pages : 223
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Book Description
This open access book provides a concise explanation of the fundamentals and background of the surround sound recording and playback technology Ambisonics. It equips readers with the psychoacoustical, signal processing, acoustical, and mathematical knowledge needed to understand the inner workings of modern processing utilities, special equipment for recording, manipulation, and reproduction in the higher-order Ambisonic format. The book comes with various practical examples based on free software tools and open scientific data for reproducible research. The book’s introductory section offers a perspective on Ambisonics spanning from the origins of coincident recordings in the 1930s to the Ambisonic concepts of the 1970s, as well as classical ways of applying Ambisonics in first-order coincident sound scene recording and reproduction that have been practiced since the 1980s. As, from time to time, the underlying mathematics become quite involved, but should be comprehensive without sacrificing readability, the book includes an extensive mathematical appendix. The book offers readers a deeper understanding of Ambisonic technologies, and will especially benefit scientists, audio-system and audio-recording engineers. In the advanced sections of the book, fundamentals and modern techniques as higher-order Ambisonic decoding, 3D audio effects, and higher-order recording are explained. Those techniques are shown to be suitable to supply audience areas ranging from studio-sized to hundreds of listeners, or headphone-based playback, regardless whether it is live, interactive, or studio-produced 3D audio material.
Author: Jean-Michel Réveillac
Publisher: John Wiley & Sons
ISBN: 1119482682
Category : Technology & Engineering
Languages : en
Pages : 381
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Book Description
For decades performers, instrumentalists, composers, technicians and sound engineers continue to manipulate sound material. They are trying with more or less success to create, to innovate, improve, enhance, restore or modify the musical message. The sound of distorted guitar of Jimi Hendrix, Pierre Henry’s concrete music, Pink Flyod’s rock psychedelic, Kraftwerk ‘s electronic music, Daft Punk and rap T-Pain, have let emerge many effects: reverb, compression, distortion, auto-tune, filter, chorus, phasing, etc. The aim of this book is to introduce and explain these effects and sound treatments by addressing their theoretical and practical aspects.
Author: Davide Rocchesso
Publisher: Mondo Estremo
ISBN: 8890112611
Category : Music
Languages : en
Pages : 244
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Book Description
Author: Philip Newell
Publisher: Routledge
ISBN: 1351371207
Category : Technology & Engineering
Languages : en
Pages : 521
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Book Description
Loudspeakers: For Music Recording and Reproduction, Second Edition is a comprehensive guide, offering the tools and understanding needed to cut out the guesswork from loudspeaker choice and set-up. Philip Newell and Keith Holland, with the assistance of Sergio Castro and Julius Newell, combine their years of experience in the design, application, and use of loudspeakers to cover a range of topics from drivers, cabinets, and crossovers, to amplifiers, cables, and surround sound. Whether using loudspeakers in a recording studio, mastering facility, broadcasting studio, film post-production facility, home, or musician’s studio, or if you simply aspire to improve your music-production system this book will help you make the right decisions. This new edition provides significant updates on the topics of digital control, calibration, and cinema loudspeaker systems.
Author: Udo Zölzer
Publisher: John Wiley & Sons
ISBN: 0470979674
Category : Computers
Languages : en
Pages : 639
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Book Description
The rapid development in various fields of Digital Audio Effects, or DAFX, has led to new algorithms and this second edition of the popular book, DAFX: Digital Audio Effects has been updated throughout to reflect progress in the field. It maintains a unique approach to DAFX with a lecture-style introduction into the basics of effect processing. Each effect description begins with the presentation of the physical and acoustical phenomena, an explanation of the signal processing techniques to achieve the effect, followed by a discussion of musical applications and the control of effect parameters. Topics covered include: filters and delays, modulators and demodulators, nonlinear processing, spatial effects, time-segment processing, time-frequency processing, source-filter processing, spectral processing, time and frequency warping musical signals. Updates to the second edition include: Three completely new chapters devoted to the major research areas of: Virtual Analog Effects, Automatic Mixing and Sound Source Separation, authored by leading researchers in the field . Improved presentation of the basic concepts and explanation of the related technology. Extended coverage of the MATLABTM scripts which demonstrate the implementation of the basic concepts into software programs. Companion website (www.dafx.de) which serves as the download source for MATLABTM scripts, will be updated to reflect the new material in the book. Discussing DAFX from both an introductory and advanced level, the book systematically introduces the reader to digital signal processing concepts, how they can be applied to sound and their use in musical effects. This makes the book suitable for a range of professionals including those working in audio engineering, as well as researchers and engineers involved in the area of digital signal processing along with students on multimedia related courses.
Author: Frank Wefers
Publisher: Logos Verlag Berlin GmbH
ISBN: 3832539433
Category : Computers
Languages : en
Pages : 278
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Book Description
This work discusses methods for efficient audio processing with finite impulse response (FIR) filters. Such filters are widely used for high-quality acoustic signal processing, e.g. for headphone or loudspeaker equalization, in binaural synthesis, in spatial sound reproduction techniques and for the auralization of reverberant environments. This work focuses on real-time applications, where the audio processing is subject to minimal delays (latencies). Different fast convolution concepts (transform-based, interpolation-based and number-theoretic), which are used to implement FIR filters efficiently, are examined regarding their applicability in real-time. These fast, elementary techniques can be further improved by the concept of partitioned convolution. This work introduces a classification and a general framework for partitioned convolution algorithms and analyzes the algorithmic classes which are relevant for real-time filtering: Elementary concepts which do not partition the filter impulse response (e.g. regular Overlap-Add and Overlap-Save convolution) and advanced techniques, which partition filters uniformly and non-uniformly. The algorithms are thereby regarded in their analytic complexity, their performance on target hardware, the optimal choice of parameters, assemblies of multiple filters, multi-channel processing and the exchange of filter impulse responses without audible artifacts. Suitable convolution techniques are identified for different types of audio applications, ranging from resource-aware auralizations on mobile devices to extensive room acoustics audio rendering using dedicated multi-processor systems.
