Using Category 5/5E/6 for Audio and Video Applications

Total Page:16

File Type:pdf, Size:1020Kb

Using Category 5/5E/6 for Audio and Video Applications Using Category 5/5e/6 for Audio and Video Applications Steve Lampen Multimedia Technology Manager Belden ©2007,©2007, Belden Belden. All rights All rights reserved. reserved. How Do We Compare? Can you use Cat 5/5e/6 for non data applications? • What applications? • What performance requirements? • Compare to TIA/EIA568-B.2 standard ©2007, Belden. All rights reserved. Table 1 – Audio Applications Application Format Cable Type Spec End-User Unbalanced Single-conductor shielded N/A Consumer Analog audio Balanced Shielded twisted-pair N/A Professional S/PDIF Consumer Unbalanced Coaxial cable Digital audio AES3-id Professional Balanced Shielded twisted-pair AES3 ©2007, Belden. All rights reserved. Table 2 – Video Applications Application Format Cable Type Spec End-User Surveillance Consumer Analog video Home video Coaxial cable Broadcast Professional Digital video Unbalanced SDI (601) S-video (dual) Y-C Pro-Sumer Analog video RGB Component Professional VGA Coaxial cable Broadband Consumer ©2007, Belden. All rights reserved. How Does a Balanced Line Work? + – ©2007, Belden. All rights reserved. How Does a Balanced Line Work? + – ©2007, Belden. All rights reserved. How Does a Balanced Line Work? + – ©2007, Belden. All rights reserved. How Does a Balanced Line Work? + – ©2007, Belden. All rights reserved. How Does a Balanced Line Work? ©2007, Belden. All rights reserved. How Does a Balanced Line Work? ©2007, Belden. All rights reserved. A Perfect Balanced Line Requirement Variations in Cable Parameter Measured in Spacing Impedance Return Loss Decibels (dB) Capacitance Capacitance Picofarads (pF) Unbalance Size Resistance Resistance Ohms (Ω) Unbalance Length Resistance Resistance Ohms (Ω) Unbalance Timing Phase Degrees ( ) “A balanced line is one where each of the two conductors, and all passive pieces attached to each conductor, are the same impedance in reference to ground.” Bill Whitlock, Jensen Transformers ©2007, Belden. All rights reserved. An Unbalanced Line • Conductors aren‟t the same size. • Conductors aren‟t the same length. • Conductors aren‟t close together. ©2007, Belden. All rights reserved. Consumer Analog Audio System Specs Category 5 Category 5e Category 6 Format Unbalanced Balanced Balanced Balanced Capacitance 30pF/ft. 15 pF/ft. 15 pF/ft. 15 pF/ft. 98 pF/m 49pF/m 49pF/m 49pF/m Impedance N/A 100Ω 100Ω 100Ω Gage 22/24 AWG (?) 24 AWG 24 AWG 23 AWG Shield YES NO NO NO ©2007, Belden. All rights reserved. How Far Can You Go? -1 dB @ 20 kHz Source 15 pF/ft. 20 pF/ft. 30 pF/ft. 50 pF/ft. Impedance (49 pF/m) (66 pF/m) (98 pF/m) (164 pF/m) 50 Ω 5406 ft. 4055 ft. 2703 ft. 1622 ft. 1648m 1236m 824m 495m 100 Ω 2707 ft. 2030 ft 1353 ft. 812 ft. 825m 619m 413m 248m 150 Ω 1873 ft. 1352 ft. 901 ft 541 ft 571m 412m 275m 165m 600 Ω 451 ft. 338 ft. 225 ft. 135 ft. 138m 103m 68.6m 41.2m 1 kΩ 271 ft. 203 ft. 135 ft 81 ft. 82.6m 61.9m 41.2m 24.7m 10 kΩ 27 ft. 20 ft 14 ft. 8 ft. 8.2m 6.1m 4.3m 2.4m 50 kΩ 5.4 ft. 4 ft. 2.7 ft. 1.6 ft. 165cm 122cm 82cm 49cm ©2007, Belden. All rights reserved. Balanced Analog Audio System Specs Category 5 Category 5e Category 6 Format Balanced Balanced Balanced Balanced 30pF/ft. 15 pF/ft. 15 pF/ft. 15 pF/ft. Capacitance 98 pF/m 49pF/m 49pF/m 49pF/m Impedance N/A 100Ω 100Ω 100Ω Gage 22-24 AWG 24 AWG 24 AWG 23 AWG Shield YES NO NO NO ©2007, Belden. All rights reserved. ©2007, Belden. All reservedrights All Belden. ©2007, Analog Audio Audio FEXT Analog in UTP 1752A FEXT Average -85 1000 1000 1000 1000 2000 2000 2000 2000 3000 3000 4000 4000 4000 4000 6000 7000 8000 8000 9000 10000 13000 14000 16000 18000 20000 24000 27000 30000 35000 39000 45000 -90 . -95 -100 Series2 dB -105 -110 -115 Frequency ©2007, Belden. All reservedrights All Belden. ©2007, Analog Audio Audio NEXT Analog in UTP 1752A NEXT AVERAGE -85 1000 1317 1634 1951 2212 2462 2712 2962 3306 3667 4043 4609 5174 5739 6412 7200 8042 8583 9300 11333 13125 14857 16833 18000 20600 24400 27667 30667 35400 40200 45500 -90 . -95 -100 dB Series1 -105 -110 -115 Frequency Balanced Pairs and Shielding • Ground Loops. – Shields can become „antennas‟. • Feed noise into the pair • Not protecting the pair from noise. • With UTP we „fixed the pair‟. – As symmetrical as possible. – As balanced as possible • No shielding to fall back on. • And now “InstaSnake” ©2007, Belden. All rights reserved. Analog Audio and MediaTwist ©2007, Belden. All rights reserved. AES3-id Professional Coax System Specs Category 5 Category 5e Category 6 Format Unbalanced Balanced Balanced Balanced Capacitance 20pF/ft. 15 pF/ft. 15 pF/ft. 15 pF/ft. 66pF/m 49pF/m 49pF/m 49pF/m Impedance 75Ω 100Ω 100Ω 100Ω Gage 20-24 AWG 24 AWG 24 AWG 23 AWG Shield YES NO NO NO Crosstalk -30 dB (?) -47.6 dB -50.6 dB -60.6 dB 6 MHz (48 kHz) PSNEXT PSNEXT PSNEXT Crosstalk -30 dB (?) -41.4 dB -41.4 dB -51.4 dB 25 MHz (192 kHz) PSNEXT PSNEXT PSNEXT Crosstalk -30 dB (?) -37.6 dB -37.6 dB -47 dB 50 MHz (384 kHz) PSNEXT PSNEXT PSNEXT ©2007, Belden. All rights reserved. What is Good CMRR? •Based on frequency SSL Mic Input Mackie 50 Hz >115 dB 1 kHz >100 dB Mic Input 10 kHz >70 dB 50 Hz >70 dB Line Input 1 kHz >70 dB 10 kHz >60 dB 50 Hz >90 dB 1 kHz >100 dB 10 kHz >50 dB ©2007, Belden. All rights reserved. What is Good CMRR? CMR of pre-amp A (GL2800) dBu -1.93 -20.00 Gain Setting 60dB 50dB -40.00 40dB 30dB -60.00 20dB 10dB -80.00 -100.00 10.00 Hz 100.00 1000.00 10000.00 30000.00 Allen & Heath GL2800 Mixer ©2007, Belden. All rights reserved. Digital Audio Sampling Rate To Determine Actual Bandwidth Bandwidth 44.1 kHz x128 5.6448 MHz 48 kHz x128 6.144 MHz 88.2 kHz x128 11.2896 MHz 96 kHz x128 12.288 MHz 176.4 kHz x128 22.5792 MHz 192 kHz X128 24.576 MHz AES5 proposed X-140 “SuperMAC” uses Category 5, 5e, 6, „7‟ ©2007, Belden. All rights reserved. S/PDIF Consumer Coax 44.1 kHz x128 = 5.6448 MHz (6 MHz) System Specs Category 5 Category 5e Category 6 Format Unbalanced Balanced Balanced Balanced Capacitance 20pF/ft. 15 pF/ft. 15 pF/ft. 15 pF/ft. 66pF/m 49pF/m 49pF/m 49pF/m Impedance 75Ω 100Ω 100Ω 100Ω Gage 23-24 AWG 24 AWG 24 AWG 23 AWG Shield YES NO NO NO Crosstalk -30 dB (?) -47.6 dB -50.6 dB -60.6 dB 6 MHz PSNEXT PSNEXT PSNEXT ©2007, Belden. All rights reserved. Balanced AES System Specs Category 5 Category 5e Category 6 Format Balanced Balanced Balanced Balanced Capacitance 13pF/ft. 15 pF/ft. 15 pF/ft. 15 pF/ft. 43pF/m 49pF/m 49pF/m 49pF/m Impedance 110Ω 20% 100Ω 15Ω 100Ω 15Ω 100Ω 15Ω Gage 24 AWG 24 AWG 24 AWG 23 AWG Shield YES NO NO NO Crosstalk -30 dB (?) -47.6 dB -50.6 dB -60.6 dB 6 MHz (48 kHz) PSNEXT PSNEXT PSNEXT Crosstalk -30 dB (?) -41.4 dB -41.4 dB -51.4 dB 25 MHz (192 kHz) PSNEXT PSNEXT PSNEXT Crosstalk -30 dB (?) -37.6 dB -37.6 dB -47 dB 50 MHz (384 kHz) PSNEXT PSNEXT PSNEXT ©2007, Belden. All rights reserved. Four, Three, Two, One-Pair • Four pair standard –3 pair (RGB) waste a pair • Two pair standard • One pair?? –Belden 1353A ©2007, Belden. All rights reserved. Analog Baseband Video System Specs Category 5 Category 5e Category 6 Format Unbalanced Balanced Balanced Balanced Capacitance 20pF/ft. 15 pF/ft. 15 pF/ft. 15 pF/ft. 77pF/m 49pF/m 49pF/m 49pF/m Impedance 75Ω 100Ω 100Ω 100Ω Gage 20-23 AWG 24 AWG 24 AWG 23 AWG Shield YES NO NO NO Crosstalk -60 dB (?) -53 dB -53 dB -63 dB 4.2 MHz PSNEXT PSNEXT PSNEXT www.belden.com “Video and UTP” ©2007, Belden. All rights reserved. Analog Video Crosstalk • Crosstalk never considered – Coax cables are shielded – Cables are random • -100 dB to –120 dB shield effectiveness • So with UTP and analog video – What crosstalk for multiple video? – Suggested: -60 dB Video Application Suggested Crosstalk Surveillance -40 dB Standard home video -50 dB Professional analog video -60 dB • A new twist with UTP: “CCTP” ©2007, Belden. All rights reserved. S-Video, S-VHS™, Y-C System Specs Category 5 Category 5e Category 6 Format Unbalanced Balanced Balanced Balanced Capacitance 17 pF/ft. 15 pF/ft. 15 pF/ft 15 pF/ft. 56 pF/m 49 pF/m 49 pF/m 49 pF/m Impedance 75Ω (?) 100Ω 100Ω 100Ω Gage 30 AWG 24 AWG 24 AWG 24 AWG Shield YES NO NO NO Crosstalk -60 dB (?) -53 dB -53 dB -63 dB 4.2 MHz PSNEXT PSNEXT PSNEXT Timing 40 nsec (?) 45 nsec 45 nsec 45 nsec (Delay Skew) (292 ft., 89m) (292 ft., 89m) (292 ft., 89m) ©2007, Belden. All rights reserved. RGB Component Video System Specs Category 5 Category 5e Category 6 Format Unbalanced Balanced Balanced Balanced Capacitance 16 pF/ft. 15 pF/ft. 15 pF/ft. 15 pF/ft. 54 pF/m 49 pF/m 49 pF/m 49 pF/m Impedance 75Ω 100Ω 15Ω 100Ω 15Ω 100Ω 15Ω Gage Wide range 24 AWG 24 AWG 24 AWG Shield YES NO NO NO Crosstalk -60 dB (?) -44.3 dB -47.3 dB -57.3 dB 10 MHz PSNEXT PSNEXT PSNEXT Timing 40 nsec 45 nsec 45 nsec 45 nsec Delay Skew (292 ft., 89m) (292 ft., 89m) (292 ft., 89m) ©2007, Belden.
Recommended publications
  • Transition to Digital Digital Handbook
    TRANSITION TO DIGITAL DIGITAL HANDBOOK Audio's video issues tion within one audio sample, the channels of audio over a fiber-optic in- There can be advantages to locking preambles present a unique sequence terface. This has since been superseded the audio and video clocks, such as for (which violate the Biphase Markby AES 10 (or MADI, Multichannel editing, especially when the audio and Code) but nonetheless are DC -freeAudio Digital Interface), which sup- video programs are related. Althoughand provide clock recovery. ports serial digital transmission of 28, digital audio equipment may provide 56, or 64 channels over coaxial cable or an analog video input, it is usually bet- Like AES3, but not fiber-optic lines, with sampling rates ter to synchronize both the audio and A consumer version of AES3 -of up to 96kHz and resolution of up to the video to a single higher -frequencycalled S/PDIF, for Sony/Philips Digi-24 bits per channel. The link to the IT source, such as a 10MHz master refer-tal Interface Format (more formallyworld has also been established with ence. This is because the former solu- known as IEC 958 type II, part of IEC- AES47, which specifies a method for tion requires a synchronization circuit60958) - is also widely used. Essen- packing AES3 streams over Asynchro- that will introduce some jitter into thetially identical to AES3 at the protocol nous Transfer Mode (ATM) networks. signal, especially because the video it- level, the interface uses consumer - It's also worth mentioning Musical self may already have some jitter. To ac- friendly RCA jacks and coaxial cable.Instrument Digital Interface (MIDI) for broadcast operations.
    [Show full text]
  • Downloadable Preview
    AES47-2006 (r2011) AES standard for digital audio — Digital input-output interfacing — Transmission of digital audio over asynchronous transfer mode (ATM) networks Published by Audio Engineering Society, Inc. Copyright ©2005 by the Audio Engineering Society Preview only Abstract This document specifies the method of carrying multiple channels of audio in linear PCM or AES3 format in calls across an asynchronous transfer mode (ATM) network to ensure interoperability. The specification includes the method of conveying information concerning the format and sampling frequency of the digital audio signal when setting up the calls. An AES standard implies a consensus of those directly and materially affected by its scope and provisions and is intended as a guide to aid the manufacturer, the consumer, and the general public. The existence of an AES standard does not in any respect preclude anyone, whether or not he or she has approved the document, from manufacturing, marketing, purchasing, or using products, processes, or procedures not in agreement with the standard. Prior to approval, all parties were provided opportunities to comment or object to any provision. Attention is drawn to the possibility that some of the elements of this AES standard or information document may be the subject of patent rights. AES shall not be held responsible for identifying any or all such patents. Approval does not assume any liability to any patent owner, nor does it assumewww.aes.org/standards any obligation whatever to parties adopting the standards document. This document is subject to periodic review and users are cautioned to obtain the latest printing. Recipients of this document are invited to submit, with their comments, notification of any relevant patent rights of which they are aware and to provide supporting documentation.
    [Show full text]
  • Maintaining Audio Quality in the Broadcast/Netcast Facility
    Maintaining Audio Quality in the Broadcast and Netcast Facility 2019 Edition Robert Orban Greg Ogonowski Orban®, Optimod®, and Opticodec® are registered trademarks. All trademarks are property of their respective companies. © Copyright 1982-2019 Robert Orban and Greg Ogonowski. Rorb Inc., Belmont CA 94002 USA Modulation Index LLC, 1249 S. Diamond Bar Blvd Suite 314, Diamond Bar, CA 91765-4122 USA Phone: +1 909 860 6760; E-Mail: [email protected]; Site: https://www.indexcom.com Table of Contents TABLE OF CONTENTS ............................................................................................................ 3 MAINTAINING AUDIO QUALITY IN THE BROADCAST/NETCAST FACILITY ..................................... 1 Authors’ Note ....................................................................................................................... 1 Preface ......................................................................................................................... 1 Introduction ................................................................................................................ 2 The “Digital Divide” ................................................................................................... 3 Audio Processing: The Final Polish ............................................................................ 3 PART 1: RECORDING MEDIA ................................................................................................. 5 Compact Disc ..............................................................................................................
    [Show full text]
  • Immersive 3D Sound Optimization, Transport and Quality Assessment Abderrahmane Smimite
    Immersive 3D sound optimization, transport and quality assessment Abderrahmane Smimite To cite this version: Abderrahmane Smimite. Immersive 3D sound optimization, transport and quality assessment. Image Processing [eess.IV]. Université Paris-Nord - Paris XIII, 2014. English. NNT : 2014PA132031. tel- 01244301 HAL Id: tel-01244301 https://tel.archives-ouvertes.fr/tel-01244301 Submitted on 17 Dec 2015 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. Université Paris 13 N◦ attribué par la bibliothèque THÈSE pour obtenir le grade de DOCTEUR DE l’UNVERSITÉ PARIS 13, SORBONNE PARIS CITÉ Discipline : Signaux et Images présentée et soutenue publiquement par Abderrahmane SMIMITE le 25 Juin, 2014 Titre: Immersive 3D sound Optimization, Transport and Quality Assessment Optimisation du son 3D immersif, Qualité et Transmission Jury : Pr. Azeddine BEGHDADI L2TI, Directeur de thèse Pr. Ken CHEN L2TI, Co-Directeur de thèse Dr. Abdelhakim SAADANE Polytech Nantes, Rapporteur Pr. Hossam AFIFI Telecom SudParis, Rapporteur Pr. Younes BENNANI LIPN-CNRS, Examinateur Mr. Pascal CHEDEVILLE Digital Media Solutions, Encadrant Declaration of Authorship I, Abderrahmane SMIMITE , declare that this thesis titled, ’Immersive 3D sound Optimization, Transport and Quality Assessment’ and the work presented in it are my own.
    [Show full text]
  • Aesjournal of the Audio Engineering Society Audio
    JOURNAL OF THE AUDIO ENGINEERING SOCIETY AES AUDIO/ACOUSTICS/APPLICATIONS VOLUME 50 NUMBER 1/2 2002 JANUARY/FEBRUARY CONTENT President’s Message........................................................................................................Garry Margolis 3 PAPERS Fourth-Order Symmetrical Band-Pass Loudspeaker Systems ....................................................................................................Grzegorz P.Matusiak and Andrzej B. Dobrucki 4 The analysis of fourth-order band-pass loudspeaker systems based on the well-known Thiele, Small, and Benson theory is presented using a reactance transformation method. By comparing the acoustic analog circuit with the transfer function obtained from the reactance transformation, the results allow the symmetry condition to be determined. The resulting system has advantages over closed-box and vented-box high-pass systems in terms of power and efficiency. Maximizing Performance from Loudspeaker Ports .......Alex Salvatti, Allan Devantier, and Doug J. Button 19 The low-frequency performance of a loudspeaker is significantly enhanced by the use of tapered ports, but there are numerous trade-offs involving the size of the port and the input and output tapers. Design issues include the effectiveness of heat transfer, amount of turbulence created, air velocity, smoothness of the taper, symmetry of the two tapers, effective mass in the port, and the contribution to the frequency response. Suggested design rules are based on extensive empirical studies. ENGINEERING REPORTS Wavetable Matching of Inharmonic String Tones ............................Clifford So and Andrew B. Horner 46 A new wavetable matching technique for inharmonic music tones, such as a violin with vibrato, shows good spectral matching and adequate frequency resolution. Confirmation with listening tests significantly improves the perceived matching but degrades performance on harmonic trumpet tones compared to simple wavetable matching.
    [Show full text]
  • Audio Sur Ip : Problématiques Et Perspectives D'évolution
    ÉCOLE NATIONALE SUPÉRIEURE LOUIS LUMIÈRE Mémoire de fin d'études AUDIO SUR IP : PROBLÉMATIQUES ET PERSPECTIVES D'ÉVOLUTION Partie Pratique : Projet d'Ingénierie d'un studio polyvalent en audio sur IP Marie-Angélique MENNECIER Directeur interne : Jean ROUCHOUSE Directeur externe : Jean-Marc CHOUVEL Rapporteur : Franck JOUANNY 19 juin 2015 RÉSUMÉ : À l'heure où les réseaux audio sur IP professionnels se développent, une normalisation a été effectuée par le groupe de travail AES X192 afin de permettre leur interopérabilité. L'AES67 pose les bases nécessaires pour un fonctionnement commun, et les principales solutions existant sur le marché de l'audio sur IP professionnel s'y sont conformées depuis. La problématique de ce mémoire est l'analyse des implications techniques de l'audio sur IP et des mécanismes nécessaires à une interopérabilité concrète, tout en proposant une ouverture sur les possibilités de développements futurs de cette technologie protéiforme. Dans un premier temps, nous formulerons quelques rappels sur les réseaux IP qui nous serviront de socle technique pour analyser les protocoles existants, puis comprendre les fondations nécessaires à leur fonctionnement commun. Nous pourrons enfin dégager quels écueils peuvent entraver de futurs développements. mots clés : AoIP, audio sur IP, AES67, Dante, Ravenna, Livewire, interopérabilité, AVB ABSTRACT : At times when audio over IP networks have started growing and thriving, a standardization has been discussed by the AES X192 workgroup to insure that they would be interoperable. In 2013, AES67 defined such rules, and the main audio over IP professional networks have since complied. This dissertation analyzes the technical details of such a technology, how interoperability can be effectively achieved, and how audio over IP may evolve, in the future, from its current protean form.
    [Show full text]
  • EBU Tech 3300S1-2005 the Middleware Report; Suppl. System
    EBU – TECH 3300s Supplement to the Middleware Report System integration in broadcast environments Geneva February 2005 Supplement to the Middleware Report Tech 3300s – E 2 Tech 3300s – E Supplement to the Middleware Report Foreword This document is the final report of the EBU project group on Middleware in Distributed programme Production (P/MDP). It is intended to be of use both to IT people working in broadcasting and to broadcasting people working with IT, and to both managerial and technical in either field. Some parts of it will seem over-simplified and some parts over-complex, but they will be different parts for different people. This is the inevitable consequence of trying to produce such a broadly accessible document. Further, it will be seen differently depending on how far along the path of 'system integration' that the reader's organisation has progressed. It might be a welcome source of information to supplement an existing development or it might be an alarming introduction to what the future holds. The EBU P/MDP Project Group was established in the autumn of 2003 to study the topic of Middleware, as the term 'Middleware' is used extensively in technical discussions but often lacks a clear definition. Triggered by this, the group started out to define Middleware terminology as well as to produce an in-depth study of the problems and the opportunities. In doing so, it became clear that what the group was really studying was 'System Integration Architectures'. The group subsequently gathered Members' experiences and held meetings with manufacturers’ representatives to discuss this topic1.
    [Show full text]
  • Audio Metering
    Audio Metering Measurements, Standards and Practice This page intentionally left blank Audio Metering Measurements, Standards and Practice Eddy B. Brixen AMSTERDAM l BOSTON l HEIDELBERG l LONDON l NEW YORK l OXFORD PARIS l SAN DIEGO l SAN FRANCISCO l SINGAPORE l SYDNEY l TOKYO Focal Press is an Imprint of Elsevier Focal Press is an imprint of Elsevier The Boulevard, Langford Lane, Kidlington, Oxford, OX5 1GB, UK 30 Corporate Drive, Suite 400, Burlington, MA 01803, USA First published 2011 Copyright Ó 2011 Eddy B. Brixen. Published by Elsevier Inc. All Rights Reserved. The right of Eddy B. Brixen to be identified as the author of this work has been asserted in accordance with the Copyright, Designs and Patents Act 1988 No part of this publication may be reproduced or transmitted in any form or by any means, electronic or mechanical, including photocopying, recording, or any information storage and retrieval system, without permission in writing from the publisher. Details on how to seek permission, further information about the Publisher’s permissions policies and our arrangement with organizations such as the Copyright Clearance Center and the Copyright Licensing Agency, can be found at our website: www.elsevier.com/permissions This book and the individual contributions contained in it are protected under copyright by the Publisher (other than as may be noted herein). Notices Knowledge and best practice in this field are constantly changing. As new research and experience broaden our understanding, changes in research methods, professional practices, or medical treatment may become necessary. Practitioners and researchers must always rely on their own experience and knowledge in evaluating and using any information, methods, compounds, or experiments described herein.
    [Show full text]
  • Low Latency Audio Processing
    Low Latency Audio Processing Yonghao Wang School of Electronic Engineering and Computer Science Queen Mary University of London 2017 Submitted to University of London in partial fulfilment of the requirements for the degree of Doctor of Philosophy Queen Mary University of London 2017 1 Statement of Originality I, Yonghao Wang, confirm that the research included within this thesis is my own work or that where it has been carried out in collaboration with, or supported by others, that this is duly acknowledged below and my contribution indicated. Previously published material is also acknowledged below. I attest that I have exercised reasonable care to ensure that the work is original, and does not to the best of my knowledge break any UK law, infringe any third party’s copyright or other Intellectual Property Right, or contain any confidential material. I accept that the College has the right to use plagiarism detection software to check the electronic version of the thesis. I confirm that this thesis has not been previously submitted for the award of a degree by this or any other university. The copyright of this thesis rests with the author and no quotation from it or information derived from it may be published without the prior written consent of the author. Signature: Date: 28/November/2017 2 Abstract Latency in the live audio processing chain has become a concern for audio engineers and system designers because significant delays can be perceived and may affect synchroni- sation of signals, limit interactivity, degrade sound quality and cause acoustic feedback. In recent years, latency problems have become more severe since audio processing has become digitised, high-resolution ADCs and DACs are used, complex processing is performed, and data communication networks are used for audio signal transmission in conjunction with other traffic types.
    [Show full text]
  • CATALOGUE ISSUE 19 Designers & Manufacturers of Audio Broadcast
    CATALOGUE ISSUE 19 Designers & Manufacturers Of Audio Broadcast Equipment Glensound Electronics Ltd. 6 Brooks Place, Maidstone, Kent, ME14 1HE, England. Tel: +44 (0)1622 753662 Fax: +44 (0)1622 762330 www.glensound.co.uk 2 Index ATM…AES47 (SONET/SDH) 8 GS1U-051 TIME CODE & REFERENCE CLOCK DISTRIBUTION 11 AMPLIFIER GS-COM001 Intercom Unit 11 BELT PACK RANGE 12 GS-FW021 Four Wire Box For Use With Headsets 12 GS-MCA001 Microphone Preamplifier 12 GS-CU004 Commentary System With Two Circuits 13 GS-HA009 Ear Piece Driver/ Headphone Amp 13 GS-HA001 Headphone Amp 13 GS-HA010 Single Input Headphone Amp 14 GS-HA011 Single Input Headphone Amplifier With Input Gain 14 GS-CU005 Two Channel Mixer with Headphone Amp 15 Adjust & Transformer Balanced Inputs GS-POT001 Headphone Pot Box 15 GS-FW008 Four Wire Talkback + Talkback into Headphone Circuit 15 GS-D2A001 Digital To Analogue Converter 16 GS-DDA001 & GS-DDA002 Two & Three Way Digital Distribution 16 Transformers GS-A2D001 Analogue to Digital Converter 16 GS-DIC001 AES3 to S/PDIF Impedance Converter 17 GS-COUGH001 Cough Switch 17 GS-OBLS-1 Loudspeaker Amplifier 17 GS-OBLS-2 Headphone & Loudspeaker Amplifier 17 GS-U2B003 Stereo Unbalanced to Balanced Converter 18 GS-PLI001 Line Ident 18 GS-CAP001 Cough & Pot Box 18 GS-SW001 Audio Switch 18 GS-MCS002 Mic Splitter (1 in 3 out) 18 GR-CM01 Roaming Compressor 19 GR-EQ01 Roaming EQ 19 COMMENTATORS’ EQUIPMENT 2 PART SYSTEMS 20 CAT5 Commentators Equipment 20 Glensound Digital Commentary: GDC-6432 21 GSOC24, GSOC26 & GSOC23 Coaxial Commentators 22 GSOC34 &
    [Show full text]
  • Presentation with Notes
    1 1 2 2 The image shows one of the original Interface Message Processors, which were used for routing packets in the Arpanet. 3 The image shows the interior of a Nine Tiles Superlink node which connected an RS422 interface into a Superlink ring network. (there was also an RS232 version, and plug-in cards for various I/O bus standards). The chipset consists of a Z80 CPU, UV-EPROM, sRAM, and an ASIC which performed a few simple functions such as converting the data between serial and parallel, and writing incoming packets directly into the sRAM. Its connection-oriented routing allowed users to specify the destination of a flow in various ways, including by name and by position in the ring. Two rings could be connected by a “gateway”, and flows could then be routed through the gateway to a destination on the other ring. Information in packet headers to identify the destination was local to each ring or gateway. Signalling was in-band, so when a connection was made to a gateway the protocol to make the connection on the second ring was carried as ordinary data on the first ring. Thus flows were source routed, reducing the amount of information each node needed to keep. When addressing content on a file server, the names of the gateway(s), host, and service simply looked like extra components in the file path. 4 The upper image is of the circuitry in the platform on which Flexilink has been implemented; it contains a Xilinx Spartan 6 SLX45T FPGA, serial flash, dRAM, and physical layer interfaces.
    [Show full text]
  • Preview - Click Here to Buy the Full Publication
    This is a preview - click here to buy the full publication IEC 62379-5-2 ® Edition 1.0 2014-03 INTERNATIONAL STANDARD Common control interface for networked digital audio and video products – Part 5-2: Transmission over networks – Signalling INTERNATIONAL ELECTROTECHNICAL COMMISSION PRICE CODE XA ICS 33.160; 35.100 ISBN 978-2-8322-1455-8 Warning! Make sure that you obtained this publication from an authorized distributor. ® Registered trademark of the International Electrotechnical Commission This is a preview - click here to buy the full publication – 2 – IEC 62379-5-2:2014 © IEC 2014 CONTENTS FOREWORD ........................................................................................................................... 5 INTRODUCTION ..................................................................................................................... 7 1 Scope ............................................................................................................................... 8 2 Normative references ....................................................................................................... 8 3 Terms and definitions ....................................................................................................... 8 4 Identification ................................................................................................................... 10 4.1 Byte order ............................................................................................................. 10 4.2 Unit identification .................................................................................................
    [Show full text]