Huron & SNAP Documentation

Huron Technical Manual 

Software version 3.2

Lake Technology Huron Technical Manual 

Software Version 3.2 

Revision 6.0 

HURON TECHNICAL MANUAL PAGE 3

Copyright © 1995-2003 Lake Technology Limited. 

First published 1995, as a Lake DSP publication. 

All rights reserved. No part of this publication may be 

reproduced or used in any form or by any means — graphic 

electronic, or mechanical including photocopying, recording, 

taping or information storage and retrieval systems — without 

written permission of the publisher. 

The enclosed materials are subject to pending and granted 

patent protection and to various patent, copyright and other 

intellectual property rights in various international jurisdictions. 

Manufactured: Sydney Australia. First printing 1995. 

Project editors: Stewart Bartlett 1995; Stephen Cox 1998; Roger 

Butler 1999-2002; Michael Potas 2002-2003. 

Section editors: Marcus Altman, Patrick Flanagan, Adam 

McKeag, Yarema Misko, Tony Place, Desmond So, Nick 

Mariette, Michael Potas. 

Page Layout/Design: Stephen Cox, Based on: “Microsoft® 

manual of style 1995 (Microsoft press)”, and “Xerox® 

publishing standards 1988 (Xerox press)”. 

InsideTrak is a registered trademark of the Polhemus 

Corporation. Lake Technology and all Lake products 

mentioned in this documentation are trademarks of Lake 

Technology Limited. Matlab is a registered trademark of 

Mathworks. Microsoft, Microsoft windows, MS, MS-DOS are 

registered trademarks of the Microsoft Corporation. Motorola 

is a registered trademark of the Motorola Corporation. Silicon 

Graphics is a registered trademark of Silicon Graphics 

Corporation. 


Disclaimer 

Copyright © 1995-2003 Lake Technology Limited. 

All rights reserved. Lake Technology Limited assumes no 

responsibility for any errors in this documentation. This 

information may supersede previously published specification 

of this product. Lake Technology Limited reserves the right to 

make changes to its products at any time without notice. 

Contact the Lake Technology Limited sales office in Sydney 

N.S.W. Australia for the latest product specification at the time 

of order. 


FCC Declaration (USA only) for 

Huron Workstation. 

Federal Communications Commission Radio Interference 

Statement Warning: 

This equipment has been tested and found to comply with the 

limits for a Class A digital device, pursuant to Part 15 of the 

FCC Rules. These limits are designed to provide reasonable 

protection against harmful interference in a commercial, 

industrial or business environment. This equipment generates, 

uses and can radiate radio frequency energy and, if not installed 

and used in accordance with the instructions, may cause 

harmful interference to radio communications. However, there 

is no guarantee that interference will not occur in a particular 

installation. If this equipment does cause harmful interference 

to radio reception, which can be determined by turning the 

equipment off and on, the user is encouraged to correct the 

interference by one or more of the following measures: 

Reorient or relocate the receiving antenna. 

Increase the separation between the equipment and the receiver. 

Connect the equipment to an outlet on a circuit different from 

that to which the receiver is connected. 

Consult the distributor or Lake Technology for help. 

Lake Technology Limited FCC Compliance Tests were 

conducted with FCC tested peripheral devices and shielded 

cables. Using equipment not FCC tested or making changes or 

modifications not expressly approved by Lake Technology 

Limited could void the users authority to operate the equipment. 


Warranty 

Lake Technology Limited A.C.N. 051 198 273 of Australia 

(“Lake”) warrants to the purchaser that the hardware and media 

embodying software in this product are free from defects in 

material and workmanship for a period of one (1) year from the 

date of purchase or shipment, whichever is earlier. 

If the purchaser discovers such a defect, they should notify 

Lake within the warranty period and return the product to Lake, 

indicating the defect and providing proof of purchase. Lake 

will examine the product and, if it finds that there is such a 

defect, will at its option either repair or replace the product, 

return the product to the purchaser within ninety (90) days of 

receipt and refund the purchaser’s shipping expenses. Shipping 

of products is at the purchaser’s sole risk. 

NOTE: — EXCLUSION AND LIMITATION OF LIABILITY 

TO THE FULL EXTENT PERMITTED BY APPLICABLE 

LAW, LAKE DISCLAIMS AND EXCLUDES ALL AND 

ANY WARRANTIES AND CONDITIONS IN RESPECT OF 

THIS PRODUCT. THIS EXCLUSION EXTENDS, WHERE 

LAWFUL TO DO SO, TO THE IMPLIED WARRANTIES OF 

MERCHANTABILITY AND FITNESS FOR A 

PARTICULAR PURPOSE. ALL SOFTWARE AND 

HARDWARE MAY HAVE ERRORS, AND THEREFORE 

THE USE OF THE LAKE SOFTWARE AND HARDWARE 

MUST BE BY SKILLED AND SUITABLY TRAINED 

PERSONNEL WHO CAN VERIFY THE RESULTS OF USE 

TO BE ACCURATE. LAKE THEREFORE SUPPLIES THE 

HARDWARE AND SOFTWARE SUPPLIED “AS IS” 

WITHOUT WARRANTY AS TO THE CORRECTNESS, 

COMPLETENESS OR ACCURACY OF THE RESULTS OF 

USE, OR AS TO ANY BENEFITS OF USE OF ANY 

NATURE. The limited warranty provided above is the only 

warranty made by Lake on this product. No oral or written 

information or advice given by Lake, its dealers, distributors, 

agents or employees will create a warranty or in any way 

increase the scope of this warranty, and you may not rely on 

any such information or advice. 

In no event will Lake be liable for any consequential, incidental 

or indirect damages (including damages for loss of business 

profits, business interruption, loss of business information) 

arising out of the use of or inability to use Lake products, even 

if Lake or an authorised Lake representative has been advised of 

the possibility of such damages or for any claim by any other 

party. 


Table of Contents 

DISCLAIMER......................................................................................................................................7 

FCC DECLARATION (USA ONLY) FOR HURON WORKSTATION.......................................9 

WARRANTY......................................................................................................................................11 

TABLE OF CONTENTS ..................................................................................................................13 

INTRODUCTION TO THE HURON .............................................................................................23 

AN OVERVIEW OF THE HURON DIGITAL AUDIO WORKSTATION ...............................25 

FEATURES .........................................................................................................................................26 

APPLICATIONS ..................................................................................................................................26 

THE HURON HARDWARE............................................................................................................27 

HURON HARDWARE COMPONENTS ...................................................................................................27 

STARTING THE HURON......................................................................................................................29 

INSTALLING NEW HARDWARE COMPONENTS ...................................................................................30 

INSTALLING AT ADD-IN CARDS .......................................................................................................31 

CONNECTING AUDIO INPUTS AND OUTPUTS TO THE HURON ......................................35 

ADDING ADDITIONAL I/O CARDS .....................................................................................................36 

THE HURON SOFTWARE COMPONENTS................................................................................37 

HURON SYSTEM TOOLS ....................................................................................................................37 

HURON SIMULATION TOOLS .............................................................................................................37 

HURON ENGINEERING TOOLS ...........................................................................................................37 

HURON PROGRAMMING TOOLS.........................................................................................................38 

LAKE SOFTWARE DESIGN SERVICE...................................................................................................38 

INSTALLING NEW SOFTWARE COMPONENTS ....................................................................39 

GENERAL INSTALLATION ..................................................................................................................39 

HURON SOFTWARE VERSION 3.2 ......................................................................................................39 

INTRODUCTION TO THE SYSTEM TOOLS .............................................................................41 

THE VRACK TOOL.........................................................................................................................43 

INTRODUCTION .................................................................................................................................43 

FEATURES .........................................................................................................................................43 

OPERATION .......................................................................................................................................44 

STARTING .........................................................................................................................................44 

CHECKING CONFIGURATION .............................................................................................................45 

CONFIGURATION FUNCTIONS............................................................................................................46 

CONFIGURATION...............................................................................................................................47 

LAUNCHING TOOLS...........................................................................................................................47 

DISPLAYING, HIDING AND TERMINATING TOOLS..............................................................................48 


TABLE OF CONTENTS 

WORKSPACES ...................................................................................................................................48 

STOPPING VRACK.............................................................................................................................50 

SUMMARY.........................................................................................................................................50 

THE PATCH BAY.............................................................................................................................51 

INTRODUCTION .................................................................................................................................51 

FEATURES .........................................................................................................................................51 

OPERATION .......................................................................................................................................51 

STARTING .........................................................................................................................................51 

PATCHING .........................................................................................................................................52 

VALID PATCHING..............................................................................................................................53 

UNPATCHING ....................................................................................................................................54 

NAMING DSP TOOLS ........................................................................................................................54 

SCROLLING .......................................................................................................................................54 

SUMMARY.........................................................................................................................................55 

THE IO MANAGER .........................................................................................................................57 

INTRODUCTION .................................................................................................................................57 

FEATURES .........................................................................................................................................57 

OPERATION .......................................................................................................................................58 

STARTING .........................................................................................................................................58 

ADJUSTING VOLUMES.......................................................................................................................59 

ADJUSTING INPUT LEVEL GAIN.........................................................................................................59 

MUTING AN OUTPUT.........................................................................................................................59 

VIEWING DIGITAL INPUT STATUS .....................................................................................................60 

SETTING DIGITAL OUTPUT FORMAT .................................................................................................60 

GROUPING VOLUME SLIDERS............................................................................................................61 

SHOWING CHANNELS IN A GROUP.....................................................................................................61 

REMOVING CHANNELS FROM A GROUP.............................................................................................62 

SPLITTING CHANNELS.......................................................................................................................62 

SPLIT WINDOWS ...............................................................................................................................62 

SUMMARY.........................................................................................................................................63 

INTRODUCTION TO THE SIMULATION TOOLS ...................................................................65 

AN OVERVIEW OF THE SIMULATION TOOLS ......................................................................67 

COORDINATE SYSTEMS.....................................................................................................................69 

CALIBRATING SPEAKER ARRAYS......................................................................................................71 

BINSCAPE .........................................................................................................................................73 

FEATURES .........................................................................................................................................73 

OPERATION .......................................................................................................................................73 

STARTING .........................................................................................................................................74 

EDITING THE LISTENER OBJECT (RECEIVER) .....................................................................................74 

ADDING AND REMOVING A SOURCE OBJECT.....................................................................................75 

EDITING A BINSCAPE OBJECT...........................................................................................................77 

ROOM SWITCHING ............................................................................................................................77 

CUSTOM ROOM FILTER SPECIFICATIONS...........................................................................................77 

CUSTOM HEAD FILTER SPECIFICATIONS ...........................................................................................78 

CREATING VALID HEAD AND ROOM FILTER FILES USING MATLAB .................................................78 

SUMMARY.........................................................................................................................................78 

THE SPACE ARRAY........................................................................................................................79 



FEATURES .........................................................................................................................................79 

OPERATION .......................................................................................................................................79 

STARTING .........................................................................................................................................80 

THE SPACE ARRAY OBJECT LIST ......................................................................................................80 

ADDING AN OBJECT TO THE SPACE ARRAY OBJECT LIST..................................................................81 

EDITING AN OBJECT..........................................................................................................................81 

REMOVING AN OBJECT......................................................................................................................82 

ADJUSTING SOURCE CHARACTERISTICS............................................................................................82 

ADDING B-FORMAT INPUT ................................................................................................................83 

SETTING MINIMUM SOURCE DISTANCE ............................................................................................83 

DEFINING THE SPEAKER ARRAY .......................................................................................................83 

Placing the Observer ...................................................................................................................84 

Adding speakers...........................................................................................................................84 

Placing speakers ..........................................................................................................................84 

Inserting speakers ........................................................................................................................85 

Deleting speakers.........................................................................................................................85 

PRE-CALCULATING SPEAKER ACTIVATIONS .....................................................................................85 

ADDING ROOM REVERBERATION......................................................................................................86 

MOVING SOUND OBJECTS.................................................................................................................86 

DSP USAGE ......................................................................................................................................86 

SUMMARY.........................................................................................................................................87 

ANISCAPE .........................................................................................................................................89 

FEATURES .........................................................................................................................................89 

OPERATION .......................................................................................................................................89 

THE SOUND FIELD FILTER.........................................................................................................91 

FEATURES .........................................................................................................................................91 

OPERATION .......................................................................................................................................91 

STARTING .........................................................................................................................................91 

EDITING THE LISTENER OBJECT (RECEIVER) ....................................................................................92 

ADDING AND REMOVING A SOUND OBJECT.......................................................................................93 

EDITING A SOUND OBJECT................................................................................................................94 

ADDING ROOM REVERBERATION......................................................................................................95 

SETTING MINIMUM SOURCE DISTANCE ............................................................................................96 

DSP USAGE ......................................................................................................................................96 

SUMMARY.........................................................................................................................................96 

THE SPEAKER DECODER ............................................................................................................97 

FEATURES .........................................................................................................................................97 

OPERATION .......................................................................................................................................97 

STARTING .........................................................................................................................................97 

ADDING, INSERTING, REMOVING AND EDITING A SPEAKER..............................................................98 

ADDING AND EDITING A LISTENER ...................................................................................................99 

TESTING SPEAKERS...........................................................................................................................99 

RESTORING A SPEAKER ARRAY ........................................................................................................99 

DSP USAGE ....................................................................................................................................100 

SUMMARY.......................................................................................................................................100 

THE BINAURAL DECODER........................................................................................................101 

FEATURES .......................................................................................................................................101 

OPERATION .....................................................................................................................................101 



STARTING .......................................................................................................................................101 

ALTERING GAIN..............................................................................................................................102 

SPECIFYING, CHANGING AND DELETING THE CHANNEL ANGLE .....................................................102 

SELECTING AND CHANGING A HEAD RELATED TRANSFER FUNCTION (HRTF) ..............................103 

SELECTING HEADPHONES ...............................................................................................................103 

DSP USAGE ....................................................................................................................................103 

SUMMARY.......................................................................................................................................104 

MULTISCAPE.................................................................................................................................105 

FEATURES .......................................................................................................................................105 

OPERATION .....................................................................................................................................105 

STARTING .......................................................................................................................................106 

ADDING EDITING AND REMOVING SOUND OBJECTS .......................................................................106 

ADDING A SOUND OBJECT ..............................................................................................................106 

EDITING A SOUND OBJECT..............................................................................................................108 

REMOVING A SOUND OBJECT..........................................................................................................108 

ADDING EDITING AND REMOVING SPEAKER ARRAYS.....................................................................108 

ADDING A SPEAKER ARRAY............................................................................................................108 

EDITING A SPEAKER ARRAY OBJECT ..............................................................................................110 

REMOVING A SPEAKER ARRAY OBJECT ..........................................................................................110 

ADDING EDITING AND REMOVING HEADPHONE OBJECTS...............................................................111 

EDITING HEADPHONE OBJECTS.......................................................................................................112 

REMOVING HEADPHONE OBJECTS ..................................................................................................112 

MULTISCAPE ROOMS......................................................................................................................112 

ROOM MAP FUNCTION....................................................................................................................113 

ADDING ROOMS, DELETING ROOMS ...............................................................................................114 

ADDING ROOM PROPERTIES............................................................................................................114 

CREATING ROOM SEGMENTS ..........................................................................................................114 

ASSOCIATING ROOM PROPERTIES WITH ROOM SEGMENTS .............................................................114 

CREATING ASSIGNING AND USING DOORS......................................................................................114 

MULTISCAPE SIMULATION OPTIONS...............................................................................................114 

NO ROOM SIMULATION ...................................................................................................................115 

ROOM SIMULATION.........................................................................................................................115 

DOOR SIMULATION .........................................................................................................................115 

DSP USAGE ....................................................................................................................................115 

SUMMARY.......................................................................................................................................116 

HEADSCAPE...................................................................................................................................117 

FEATURES .......................................................................................................................................117 

OPERATION .....................................................................................................................................117 

STARTING .......................................................................................................................................118 

SETTING THE ANGLE CHANNEL ......................................................................................................118 

ADDING THE ENVIRONMENT RELATED TRANSFER FUNCTIONS (ERTF’S) ......................................118 

ADDING, EDITING AND DELETING EARLY IMPULSES ......................................................................119 

CONFIGURING HEADSCAPE.............................................................................................................120 

SAVING ...........................................................................................................................................120 

HEADSCAPE FILES AND THE HEADSCAPE FILE FORMAT .................................................................120 

DSP USAGE ....................................................................................................................................121 

SUMMARY.......................................................................................................................................122 

SONIC ANIMATOR .......................................................................................................................123 

FEATURES .......................................................................................................................................123 



OPERATION .....................................................................................................................................123 

STARTING .......................................................................................................................................125 

MANIPULATING TRACKS.................................................................................................................128 

MANIPULATING TRAJECTORIES.......................................................................................................129 

SEQUENCING FUNCTIONS................................................................................................................130 

EXTERNAL MTC SYNCHRONISATION .............................................................................................131 

USER INTERFACE FEATURES ...........................................................................................................131 

DSP USAGE ....................................................................................................................................131 

SUMMARY.......................................................................................................................................131 

OTHER SIMULATION TOOLS ...................................................................................................133 

THE LOCATOR..............................................................................................................................135 

FEATURES .......................................................................................................................................135 

OPERATION .....................................................................................................................................136 

STARTING .......................................................................................................................................136 

SETTING A SYMBOL TO REPRESENT A SOUND OBJECT OR LISTENER OBJECT...................................136 

MOVING A LISTENER OBJECT OR SOUND OBJECT USING THE LOCATOR CONTROL PANEL..............137 

LOCATOR SCRIPTS ..........................................................................................................................137 

IMPORTANT INFORMATION CONCERNING THE WRITING OF LOCATOR SCRIPTS: .............................137 

LOCATOR COMMANDS ....................................................................................................................138 

LOADING AND PLAYING LOCATOR SCRIPTS....................................................................................138 

PAUSING AND RESTARTING THE LOCATOR PLAYBACK ...................................................................138 

RECORDING SCRIPTS.......................................................................................................................138 

SUMMARY.......................................................................................................................................139 

THE WAVEPLAYER-2..................................................................................................................141 

FEATURES .......................................................................................................................................141 

REQUIREMENTS...............................................................................................................................141 

HARDWARE CONNECTION...............................................................................................................142 

DRIVER INSTALLATION ...................................................................................................................142 

OPERATION .....................................................................................................................................142 

STARTING .......................................................................................................................................144 

SETTING THE SYSTEM SAMPLE RATE..............................................................................................144 

SYSTEM LATENCY...........................................................................................................................144 

SETTING CLOCK SYNCHRONISATION ..............................................................................................144 

WAVEPLAYER-2 AUDIO OUTPUTS IN THE HURON PATCHBAY........................................................145 

CONTROLLING WAVEPLAYER-2 .....................................................................................................145 

OPTIMISING SYSTEM PERFORMANCE ..............................................................................................146 

SUMMARY.......................................................................................................................................146 

THE WAVEPLAYER .....................................................................................................................147 

FEATURES .......................................................................................................................................147 

OPERATION .....................................................................................................................................147 

STARTING .......................................................................................................................................148 

CONFIGURING VOICES ....................................................................................................................148 

LOADING, PLAYING, STOPPING AND CONTROLLING WAVEPLAYER ...............................................148 

SUMMARY.......................................................................................................................................149 

MAKE ROOM .................................................................................................................................151 

FEATURES .......................................................................................................................................151 

OPERATION .....................................................................................................................................152 

STARTING .......................................................................................................................................152 



SAVING A MAKE ROOM SETUP .......................................................................................................157 

RESTORING A MAKE ROOM SETUP..................................................................................................157 

SUMMARY.......................................................................................................................................157 

SPATIAL NETWORK AUDIO PROTOCOL (SNAP)................................................................159 

INTEGRATION UNDER 32 BIT WINDOWS..........................................................................................159 

INTEGRATION UNDER UNIX.............................................................................................................159 

API .................................................................................................................................................159 

INITIALISATION...............................................................................................................................160 

SNAP COMMANDS .........................................................................................................................160 

General VRack Commands........................................................................................................160 

Positional Commands ................................................................................................................161 

AniScape Commands .................................................................................................................162 

MultiScape Commands ..............................................................................................................165 

Space Array Commands.............................................................................................................167 

VMixer Commands.....................................................................................................................169 

WavePlayer-2 Commands..........................................................................................................171 

WavePlayer Commands .............................................................................................................174 

COMMAND REFERENCE............................................................................................................177 

LOCATOR COMMANDS ....................................................................................................................178 

INTRODUCTION TO THE ENGINEERING TOOLS...............................................................183 

THE CONVOLVER TOOL............................................................................................................185 

INTRODUCTION ...............................................................................................................................185 

FEATURES .......................................................................................................................................185 

OVERVIEW OF THE CONVOLVER SETUP ..........................................................................................185 

OPERATION .....................................................................................................................................187 

STARTING AND SAVING...................................................................................................................187 

SELECTING THE FILTER...................................................................................................................187 

CREATING PROGRAMS ....................................................................................................................188 

INPUTS, OUTPUTS, MIXING AND PATCHING....................................................................................190 

INPUT AND OUTPUT LEVEL METERING ...........................................................................................191 

FAST SWITCHING WITH PRE-LOADED FILTER COEFFICIENTS...........................................................191 

SUMMARY.......................................................................................................................................192 

THE 96KHZ CONVOLVER TOOL..............................................................................................193 

INTRODUCTION ...............................................................................................................................193 

96KHZ CONVOLVER HARDWARE REQUIREMENTS ..........................................................................193 

96KHZ FILTERS...............................................................................................................................194 

PATCHING THE 96KHZ CONVOLVER ...............................................................................................195 

LOADING THE 96KHZ CONVOLVER.................................................................................................195 

THE MEASUREMENT TOOL......................................................................................................197 

INTRODUCTION ...............................................................................................................................197 

FEATURES .......................................................................................................................................197 

OPERATION .....................................................................................................................................197 

STARTING .......................................................................................................................................198 

SELECTING THE TEST SIGNAL .........................................................................................................198 

GENERATING THE OUTPUT SIGNAL.................................................................................................198 

INPUTS ............................................................................................................................................199 

SUM MULTIPLIER............................................................................................................................199 



DC BLOCK......................................................................................................................................199 

SELECTING THE OUTPUT FILE .........................................................................................................200 

RUNNING THE MEASUREMENT........................................................................................................200 

LEVEL METERING ...........................................................................................................................200 

GENERATING TEST SIGNALS ...........................................................................................................201 

NOTES ON TEST SIGNALS................................................................................................................201 

SIGNAL GENERATION VIA MATLAB SCRIPTS ..................................................................................202 

SUMMARY.......................................................................................................................................203 

THE 96KHZ MEASUREMENT TOOL........................................................................................205 

INTRODUCTION ...............................................................................................................................205 

96KHZ MEASUREMENT TOOL’S HARDWARE REQUIREMENTS ........................................................205 

PATCHING THE 96KHZ MEASUREMENT TOOL.................................................................................206 

LOADING THE 96KHZ MEASUREMENT TOOL ..................................................................................207 

INTRODUCTION TO THE SIM TOOLS....................................................................................209 

THE SIM FILE FORMAT .............................................................................................................211 

DATA FORMATS..............................................................................................................................211 

THE SIMWIN TOOL......................................................................................................................213 

INTRODUCTION ...............................................................................................................................213 

OPERATION .....................................................................................................................................213 

STARTING .......................................................................................................................................213 

SELECTING AND EXECUTING A FUNCTION ......................................................................................214 

SIMWIN — DFILTER TOOL .......................................................................................................215 

DFILTER WINDOWS INTERFACE ......................................................................................................215 

SIMWIN — LVEC TOOL..............................................................................................................217 

LVEC WINDOWS INTERFACE ...........................................................................................................217 

SIMWIN — PADVEC TOOL ........................................................................................................219 

PADVEC WINDOWS INTERFACE.......................................................................................................219 

SIMWIN — CUTVEC TOOL ........................................................................................................221 

CUTVEC WINDOWS INTERFACE.......................................................................................................221 

SIMWIN — PERMUTE TOOL.....................................................................................................223 

PERMUTE WINDOWS INTERFACE.....................................................................................................223 

SIMWIN — WINGEN TOOL........................................................................................................225 

WINGEN WINDOWS INTERFACE ......................................................................................................225 

SIMWIN — SIGGEN TOOL .........................................................................................................227 

SIGGEN WINDOWS INTERFACE .......................................................................................................227 

SIMWIN — VECTOR ARITHMETIC TOOL ............................................................................231 

VECTOR ARITHMETIC WINDOWS INTERFACE..................................................................................231 

THE PLOTWSIM TOOL ...............................................................................................................233 

INTRODUCTION ...............................................................................................................................233 

FEATURES .......................................................................................................................................233 

OPERATION .....................................................................................................................................233 



READING SIM FILES .......................................................................................................................234 

PRINTING ........................................................................................................................................235 

COPYING THE GRAPH TO THE CLIPBOARD........................................................................................236 

VIEWING THE DATA........................................................................................................................236 

ALTERING THE AXES.......................................................................................................................237 

SUMMARY.......................................................................................................................................239 

THE SIMSCALE TOOL.................................................................................................................241 

INTRODUCTION ...............................................................................................................................241 

FEATURES .......................................................................................................................................241 

OPERATION .....................................................................................................................................241 

SUMMARY.......................................................................................................................................242 

WAVTOSIM ....................................................................................................................................243 

USAGE ............................................................................................................................................243 

SIMTOWAV ....................................................................................................................................245 

USAGE ............................................................................................................................................245 

FREQUENCY RESPONSES FOR WINDOW FUNCTIONS — SIMTOOLS APPENDIX 1 247 

INTRODUCTION ...............................................................................................................................247 

UNIFORM WINDOW.........................................................................................................................247 

BARTLETT WINDOW .......................................................................................................................247 

VON HANN WINDOW ......................................................................................................................247 

HAMMING WINDOW........................................................................................................................248 

BLACKMAN WINDOW .....................................................................................................................248 

HARRIS WINDOW............................................................................................................................248 

BLACKMAN-HARRIS WINDOW........................................................................................................249 

INTRODUCTION TO THE PROGRAMMING TOOLS ...........................................................251 

INTRODUCTION TO EQUALISATION TOOLS......................................................................253 

FEATURES .......................................................................................................................................253 

APPLICATIONS ................................................................................................................................253 

OPERATION .....................................................................................................................................254 

STARTING LAKE EQ........................................................................................................................254 

DESCRIPTION OF THE EQ INTERFACE..............................................................................................254 

CREATING AN EQ CANVASS ...........................................................................................................256 

LOADING EQ CANVASSES ..............................................................................................................257 

CLICKING AND DROPPING SLIDER WIDGETS...................................................................................257 

CALCULATING THE EQUALISATION CURVE.....................................................................................260 

BYPASS THE EQUALISATION CURVE ...............................................................................................260 

SETTING THE EQ FILTER LENGTH...................................................................................................260 

LOCKING SLIDERS...........................................................................................................................261 

MODIFYING OVERALL CUT/BOOST.................................................................................................262 

ZOOMING IN AND OUT ....................................................................................................................262 

SAVING ...........................................................................................................................................263 

LIMITS ............................................................................................................................................264 

PHASE CONSIDERATIONS ................................................................................................................264 

EXPORTING AND IMPORTING EQUALISATION DATA........................................................................265 

STOPPING EQ..................................................................................................................................265 

SUMMARY.......................................................................................................................................265 



THE 96KHZ EQ ..............................................................................................................................267 

INTRODUCTION ...............................................................................................................................267 

96KHZ EQ HARDWARE REQUIREMENTS.........................................................................................267 

96KHZ FILTERS...............................................................................................................................268 

PATCHING EQ 96KHZ.....................................................................................................................268 

LOADING EQ 96KHZ ......................................................................................................................269 

OTHER TOOLS ..............................................................................................................................271 

THE VMIXER .................................................................................................................................273 

FEATURES .......................................................................................................................................273 

OPERATION .....................................................................................................................................273 

STARTING .......................................................................................................................................274 

OPENING AND SAVING SETTINGS....................................................................................................274 

RESIZING.........................................................................................................................................274 

LABELLING CHANNELS ...................................................................................................................275 

ROUTING AUDIO .............................................................................................................................275 

ADJUSTING LEVEL ..........................................................................................................................276 

GROUPING LEVEL CONTROLS .........................................................................................................276 

MUTING ..........................................................................................................................................277 

DISPLAYING LEVEL METERS...........................................................................................................277 

EXITING VMIXER............................................................................................................................277 

DSP USAGE ....................................................................................................................................277 

CONTROLLING VMIXER REMOTELY ...............................................................................................278 

SUMMARY.......................................................................................................................................278 

ABSWITCHER................................................................................................................................279 

OPERATION .....................................................................................................................................279 

STARTING .......................................................................................................................................279 

SELECTING AN OUTPUT PAIR ..........................................................................................................279 

DISPLAYING METERS......................................................................................................................280 

EXITING AB SWITCHER ..................................................................................................................280 

SUMMARY.......................................................................................................................................280 

BFORMAT MIXER ........................................................................................................................281 

OPERATION .....................................................................................................................................281 

STARTING .......................................................................................................................................281 

ALTERING GAIN LEVELS.................................................................................................................281 

ALTERING THE XYZ BALANCE.......................................................................................................282 

ALTERING THE AZIMUTH ................................................................................................................282 

ALTERING THE ELEVATION.............................................................................................................282 

VIEWING BFORMAT MIXER METERING ..........................................................................................282 

SUMMARY.......................................................................................................................................283 

I/O TEST AND I/O 96K TEST.......................................................................................................285 

OPERATION .....................................................................................................................................285 

STARTING .......................................................................................................................................285 

I/O TEST PATCHING........................................................................................................................286 

SELECTING THE FORMAT OF RESULTS.............................................................................................286 

TESTING SIGNAL TO NOISE RATIOS ................................................................................................287 

TESTING I/O’S FOR DEVIATION FROM NORMAL FREQUENCY CHARACTERISTICS............................287 

SUMMARY.......................................................................................................................................288 



HURON TECHNICAL REFERENCE..........................................................................................289 

DATA FORMATS AND SCALING CONSIDERATIONS.........................................................291 

FIXED POINT DIGITAL AUDIO DATA REPRESENTATION ..................................................................291 

FINITE IMPULSE RESPONSE FILTER (FIR) COEFFICIENT SCALING...................................................291 

HURON TECHNICAL SPECIFICATIONS ................................................................................295 

PERFORMANCE PLOTS.....................................................................................................................295 

MECHANICAL SPECIFICATIONS .......................................................................................................298 

AUDIO OUT CONNECTOR (MALE) ....................................................................................................298 

AUDIO IN CONNECTOR (FEMALE) ....................................................................................................298 

D-62 CONNECTOR SPECIFICATIONS ................................................................................................299 

MANAGING AUDIO SIGNAL LEVELS FOR OPTIMUM NOISE PERFORMANCE ........301 

AUDIO INPUT LEVELS .....................................................................................................................301 

AUDIO OUTPUT LEVELS..................................................................................................................302 

CONNECTING OUTPUTS TO A POWER AMPLIFIER WITH GAIN CONTROL..........................................303 

GLOSSARY......................................................................................................................................305 

APPENDIX I: AUTO-STARTING THE HURON ON POWER-UP. ........................................311 

VRACK AUTOSTART .......................................................................................................................311 


Introduction to the Huron 

Thank you for purchasing the Huron Digital Audio Convolution 

Workstation. 

This manual is organised in sections each of which focus on 

specific software packages purchased with your system. The 

Huron software and hardware have been preinstalled in your 

system, however, we recommend reading the following 

“Introduction to Huron” to familiarise yourself with the 

installation and upgrading of Huron software and hardware. 

We also recommend reading the Readme.txt and 

Release.txt files found on the CD supplied with your 

system, for the latest information concerning the software 

release. 


An Overview of the Huron Digital 

Audio Workstation 

Figure 1 — The Huron system 

The Huron Digital Audio Convolution Workstation is an 

extremely powerful and modular general purpose signal 

processing engine for professional audio systems. It is designed 

around a 256 channel digital audio data bus. Interfacing with 

this bus are a combination of Huron Digital Signal Processing 

(DSP) boards and Huron Input and Output (I/O) boards. This 

specialised hardware is controlled by an industry standard PC 

host system. 

Huron specialises in — but is not limited to — applications 

which demand real time convolution, such as complex audio 

simulation and virtual reality audio. The Huron may be 

configured to process applications that require from two 

channels to more than 64 channels. 

Lake Software provides a simple graphical interface through 

which it is possible to run a variety of Lake designed audio 

engineering tools. Additionally, Lake provides a C++ Library 

interface for custom application programming. 


Features 

Applications 

Note: Any ISA or PCI card may be 

added to Huron workstations 

• Modular DSP and I/O architecture. 

• Internal 256 channel digital audio bus. 

• High performance parallel DSP architecture. 

INTRODUCTION TO THE HURON 

• Massive convolution with zero propagation delay. 

• Fast animating convolution. 

• General purpose, programmable DSP's, with zero wait state 

DRAM for memory intensive applications. 

• Professional quality analogue or digital audio interfaces. 

• Object oriented programming interface, with dynamic link 

libraries and DSP tools. 

• TCP/IP control from external workstations such as Silicon 

Graphics ® machines. 

• Virtual reality audio. 

• Animated auralisation, virtual walk-through acoustics. 

• Multi-speaker audio presentations. 

• Audio simulation and synthesis. 

• Room measurement and equalisation. 

• Multi-effects for audio post production. 

• Forensic audio. 

• Integrated sound reinforcement systems. 

• Audio workstation network. 


Huron Hardware Components 

The Huron Hardware 

The Huron is a stand-alone audio workstation which consists of 

the following components: 

• Chassis: The Huron Workstation comes in a chassis painted 

to specification. The chassis contains a high quality 

industrial power supply, cooling fans with filters, a 3.5 inch 

floppy disk drive, a CD-ROM drive and hard disk drives 

ordered to the user’s specification. 

• Huron Backplane: The standard Huron backplane contains 

four PCI bus connectors, seven ISA bus connectors, a dual 

PCI-ISA edge connector and seven 256 channel Huron bus 

connectors. The Huron 20 backplane consists of sixteen 16 

bit ISA bus connectors, two 32 bit PCI connectors, a Dual 

PCI-ISA edge connector and sixteen 256 channel Huron bus 

connectors. 

• Single Board IBM PC Compatible Computer with 

Embedded Video and an Ethernet Network Card, supporting 

RJ-45 and Coaxial BNC connectors. 

The Huron also consists of a user specified number of: 

• DSP Cards: The Lake Technology Card contains four 

Motorola® 56002 DSP processors, and Lake proprietary 

circuitry. 

• ADAT I/O cards: The ADAT I/O card has three ADAT 

optical input and output connectors providing a total of 24 

channels of digital I/O. 

• I/O carrier cards: The I/O carrier card has connectors to 

support up to 8 plug-in modules. It interfaces the I/O 

modules to the Huron bus. 

• 16 bit dual channel A/D Module. 

• 18 bit dual channel D/A Module. 

• AES/EBU Synchronous Digital Input Module. 

• AES/EBU Asynchronous Digital Input Module. 

• AES/EBU Digital Output Module. 

• 16 channel Patch Panel. 

• RME Hammerfall card for WavePlayer-2. This card 

provides 24 channels of ADAT optical I/O for interfacing 

with Huron ADAT I/O cards and other third party digital 

audio equipment. 



• Optional ISA cards. Any ISA bus card may be optionally 

incorporated into the Huron system. 

• Optional PCI cards. Any PCI cards may be optionally 

incorporated into the Huron system. 

The following diagrams illustrate the Huron chassis (Figure 2 

and Figure 3). The system illustrated has a Huron ADAT I/O 

card in slot 0, a DSP card in slot 1, an I/O carrier card in slot 2, 

and a RME Hammerfall digital audio PCI card for use with the 

WavePlayer-2 application. 

Figure 2 — Front elevation of the Huron. 

1 Reset switch 14 Parallel port connector 

2 Keyboard lock 15 SVGA port Connector 

3 Power indicator 16 Mouse connector 

4 Hard disk indicator 17 COM1 connector 

5 Keyboard lock indicator 18 Keyboard connector 

6 Power switch 19 I/O connector 

7 Floppy disk drive 20 Huron ADAT I/O connectors 

8 CD-ROM Drive 21 Hammerfall SPDIF I/O and ADAT Sync connector 

9 Power Supply Unit 22 Hammerfall ADAT1 and ADAT2 I/O connectors 

10 Mains inlet 23 Hammerfall ADAT3 I/O connectors 

11 Mains outlet 24 Hammerfall Word Clock out connector 

12 Ethernet connector 25 Hammerfall Word Clock in connector 

13 COM2 connector 


↓ Setting up the Hardware 

Starting the Huron 

↓ To start the Huron workstation 

(see Figure 3) 

Figure 3 — Rear elevation of the Huron. 


You will need the following hardware before you can use the 

Huron: 

• 1 PS2 keyboard or AT keyboard with PS2 adaptor plug. 

• 1 PS2 Microsoft mouse (or compatible). 

• 1 SVGA screen with IEC mains extension lead. 

• 1 IEC mains lead. 

1. Connect the keyboard cable (with the adaptor if required) to 

the keyboard socket (18). 

2. Connect the mouse cable to the mouse socket (16). 

3. Connect the SVGA cable to the SVGA port (15). 

4. Plug the mains extension lead into the mains outlet of Huron 

5. 11) and the mains inlet of the SVGA screen. 

6. Plug the mains cable into the Huron mains inlet (10). 

7. Unlock the front panel drive bay with the key provided. The 

disk drive and CD ROM drive should be empty. 

8. Turn on the SVGA screen then turn the power switch on. 

After an initial boot process, log onto Windows 2000. 


Installing New Hardware Components 

Note: Always observe anti-static 

precautions when handling electronic 

equipment. 

↓ Installing new DSP or I/O cards 

↓ Installing new I/O modules 


New Huron cards and modules can be purchased to expand the 

capability of Huron. Qualified service personnel should 

perform the installation of these cards or modules. 

These guidelines should be followed when installing new cards: 

Addition and removal of all cards should only be attempted 

when the power to the Huron system is off. When installing a 

Huron ADAT I/O card, this should be installed in slot 0 (the 

leftmost slot when viewing the Huron from the front). When 

installing a new DSP or I/O carrier card, install it in the slot 

directly next to the last Huron DSP or I/O card. There should 

be no vacant slots between Huron DSP and I/O cards. 

Addition and removal of all modules should only be attempted 

when the power to the Huron system is off. When installing 

new I/O modules onto a Huron I/O carrier card, the position and 

orientation of the modules is very important. Input modules are 

always populated in module 8 first, followed by module 7, and 

so on. Output modules are always populated in module 1 first, 

followed by module 2, and so on. When inserting a new input 

module insert it in the highest available slot number. When 

inserting a new output module, insert it in the lowest available 

slot number. Always be careful to ensure the correct orientation 

of any modules. The orientation shown in Figure 4 must be 

adhered to for all modules. Special attention must also be paid 

to ensuring the module connector pins mate perfectly with the 

I/O base card connectors. Incorrect insertion of the modules 

could result in damage to the module and/or the workstation. 


Installing AT Add-In Cards 

If the Huron System does not work 

after the installation of an ISA card. 

↓ Check the following settings 

Figure 4 — Correct orientation of I/O module during installation 


The Huron system is an AT compatible computing system. 

Standard ISA and PCI cards can be added to the system and 

configured for correct operation. The installation manuals for 

any add-in cards should be followed carefully to ensure their 

correct operation in the system. 

If the Huron system does not work after adding an AT 

compatible card to the system you should check the 

configuration of Huron and the add-in card. The most common 

problem is a conflict between an I/O ROM of an ISA add-in 

card and the address space used by the Huron system. 

• The Huron system has a floppy controller built in to the 

single board computer. Some SCSI cards have a floppy disk 

drive controller included. This additional SCSI drive 

controller should be disabled. 

• Check the base address of any RAM or I/O ROM on the AT 

add-in card. Some typical default base addresses for add in 

cards are 0xC800 for ethernet cards and 0xDC00 for SCSI 

cards. 

If you determine that the base addresses for the Huron system 

and the new add-in card overlap, then you must change one of 

the base addresses. Either follow the instructions in the 

installation manual of the new add-in card or change the Huron 

base address as explained below. 


↓ To change the ISA memory 

address of the Huron 

↓ To edit the registry settings 


Altering the ISA base memory address of the Huron is a two 

stage process, it requires that: 

1. The Jumpers on the DSP card and I/O card are set to a new 

address. 

2. The Huron Windows 2000 Registry settings are altered to 

refer to the changed addresses. 

• Adjust the jumpers on the DSP card. 

JP2 is located on the bottom left hand corner of the DSP card. 

The base address can be located on any 0x0200 boundary, but 

should be confined to 0xD000-0xDE00 for AT compatible 

systems. The most significant 7 bits of the base address are set 

by JP2, with the most significant bit (MSB) on the bottom of 

the jumper block. When no jumper is installed on a bit position 

a logical “1” is set, while an installed jumper denotes a logical 

“0”. 

Below is a diagram of the jumper setting for the DSP card 

default base address of 0xD000. If you have any other cards in 

your computer, ensure they do not access the same address 

space as the Huron. 

Figure 5 — DSP card Jumper 2 

• Set the I/O carrier card to the same address as the DSP Card 

using JP 1 on the I/O carrier card. 

Below is a diagram of the I/O card jumper settings for the 

default base address of 0xD000. 

Figure 6 — I/O card Jumper 1 

After the Huron hardware base address has been changed, the 

Huron Windows 2000 Registry settings need to be edited to 

match this new base address. 

1. From the Windows 2000 Start Menu select Run. 

2. At the prompt type Regedit or Regedt32. 

3. Browse to the Registry Key 

HKEY_LOCAL_MACHINE\SYSTEM\CurrentControl 

Set\Services\lakekrnl\Parameters 



4. Edit the IoMemoryMapBase value to the new values 

previously set on the card. 

5. Eg. 0xd0000 (the default address) 


↓ Connecting with the patch panel 

Connecting Audio Inputs and 

Outputs to the Huron 

Audio inputs and outputs are connected to the Huron system via 

1U 19” rack-mount patch panels or ADAT optical connectors 

located on the Huron ADAT I/O card. 

Each patch panel is wired with 16 XLR connectors and one D62 

male connector. The gender of the XLR connectors used is 

tailored to the configuration of the I/O card with which the 

panel is intended to mate. Male connectors are used for outputs 

and female connectors are used for inputs. A typical example is 

shown in Figure 7. This panel is to mate with an I/O card with 

1-4 output modules and 1-4 input modules. 

Each IO module in the Huron will consume 2 input or output 

points on the patch panel regardless of whether they are 

analogue or digital. 

An analogue IO will output or take as input both left and right 

channels through two points on the connector panel, requiring 

two XLR connectors. For the example below these are 

represented by output points 1 and 2 and input points 13 and 14. 

A digital IO will output or take inputs for both the left and right 

channels though a single point on the connector panel, requiring 

only a single XLR connection. The input or output to the 

digital IO should be on the odd numbered connector. In the 

example below this is represented by the digital input 15 (the 

number 16 point is skipped or ignored). 

Figure 7 — The relationship between software and physical patch points 


↓ Connecting ADAT I/O 

Adding Additional I/O Cards 


Each ADAT connection on the Huron ADAT I/O card (shown 

in Figure 3) carries 8 channels of digital audio data via optical 

TOSLINK connectors. When viewing the Huron from the rear, 

the first (top), third and fifth connectors are for audio inputs 1- 

8, 9-16, and 17-24 respectively, and the second, fourth and sixth 

(bottom) connectors are for audio outputs 1-8, 9-16, and 17-24 

respectively. 

If a second I/O carrier card is installed in the system of Figure 7 

its channels would be numbered from 17 to 32. To determine 

which physical patch panel inputs and outputs belong to those 

shown in the configuration window, the configuration numbers 

should be converted to modulo 16. 

For further information regarding the pin outs of the patch panel 

connectors see Huron Hardware Specifications. 


Huron System Tools 

Huron Simulation Tools 

Huron Engineering Tools 

The Huron Software Components 

Huron software is distributed in the following packages. 

The Huron System Tools package forms the basis of the Huron 

system software. Central to the System Tools is the virtual rack 

or VRack. The VRack provides a standard interface through 

which users are able to load various signal processing 

applications. The System Tools also include a Patch Bay for 

connecting application modules and physical inputs and 

outputs, and an Input Output Manager for controlling gain and 

the configuration of hardware. 

Lake Technology’s simulation tools are designed for use in 

audio simulations. Designed to complement Virtual Reality 

applications, the simulation tools can be chosen for either arrays 

of speakers or headphones. Tools such as Space Array and 

Binscape are provided as an interface through which sound 

sources and virtual Soundscapes may be controlled via TCP/IP. 

Additional tools provide access to sound playback and effects 

such as speaker panning. 

The Huron Engineering Tools are designed for audio research 

purposes. Applications include: a Convolver which allows the 

use of long low latency Finite Impulse Response (FIR) filters; a 

Measurement tool which provides a means of recording the 

impulse response of linear systems (such as audio processors 

and large acoustic spaces); and a set of tools for plotting 

generating and formatting data files in the SIM file format used 

by all Lake software. 


Huron Programming Tools 

Lake Software Design Service 


Lake provides the Programming Tools for customers who wish 

to design their own specific DSP and application programs. A 

C++ DSP Programmers Library is provided, allowing 

customers to develop their own DSP algorithms and 

applications on the Huron platform. Alternatively, C++ classes 

are provided which enable application programmers to 

incorporate Lake developed algorithms into custom-built 

applications. Version 3.2 of the programming tools includes a 

Matlab® interface which enables Matlab 5.2 or 5.3 (32 bit 

versions) users to compose and run scripts which utilise the 

Huron hardware for the Measurement and Convolution of audio 

data. 

Lake also offers a design contract service to build specific 

Huron applications to meet customer requirements. Contact 

Lake Technology for more information on custom designed 

software. 


General Installation 

↓ To install the Huron license 

↓ To install Huron VRack 

software. 

Huron Software Version 3.2 

Installing New Software Components 

Software components for the Huron come on a CD. Along with 

the CD is a licence floppy disk, which “enables” the software 

components you have purchased. 

Each of the software components supplied for Huron are 

installed from Windows 2000 in the same way: the order of 

installation must be the Software Licence first, the Huron 

Software second, and any upgrades or patches last. 

1. Start Windows 2000 

2. Insert the Licence diskette into drive A:\ 

3. Click the Start button, then select Run…. When the Run 

dialogue box appears, enter A:\setup where, A:\ is the drive 

letter of the Floppy Disk Drive. 

Follow the on screen instructions to complete the license 

installation. 

1. Start Windows 2000 

2. Insert the Installation CD in the CD-ROM drive. 

3. Click the Start button, then select Run…. When the Run 

dialogue box appears, enter [CD]:\setup where [CD] is the 

drive letter of your CD-ROM drive. 

4. Click OK. 

Follow the instructions that appear on the screen. 

The Huron Software installation will modify your Huron’s 

configuration files. 

After the installation, the Huron needs to be restarted for these 

changes to take effect. A dialogue box will appear to confirm 

the restart of the computer. 


↓ Changes to the Windows 2000 

Environment 

↓ Changes to Path Variable 

↓ Update the Windows 2000 

Registry 

↓ When using Matlab 

Programming Tools 


The following environment variable will be added to your 

Windows 2000 Environment Variable List if you choose the 

default installation directory of C:\Huron32. 

LAKE-CFG C:\Huron32 

The following path will be added to your Windows 2000 Path 

Environment Variable if you choose the default installation 

directory of C:\Huron32. 

PATH C:\Huron32\Bin 

The Huron 3.2 installation will also update information in the 

Windows 2000 Registry. 

When using the Matlab programming tools it is important that 

the Windows 2000 path environment variable includes the 

Matlab binary directory eg. C:\Matlab\Bin 

The Matlab application should also include in its path variable 

the following: 

C:\Huron32\Bin 

C:\Huron32\Huronmat 

C:\Huron32\Examples\Matlab 


Introduction to the System Tools 

The System tools serve as the framework for all Huron software 

applications. They provide a standard interface for: 

• Configuring the physical input and output modules; 

including the naming of signals, the setting of input and 

output volumes and the manipulation of preamplifier gains. 

• Digitally connecting these physical inputs and outputs to the 

inputs and outputs of DSP application programs. 

• Starting and stopping applications, and controlling their onscreen 

visibility. 

The heart of the system software is the Virtual Rack or VRack. 

The VRack provides a graphical user interface based upon the 

metaphor of a “19-inch audio rack”. All DSP applications are 

available to the user as VRack mountable units. Each of these 

units may contain a number of input and output connectors. 

To connect the physical inputs and outputs to these DSP 

application input and output connectors, the Patch Bay is used. 

The Patch Bay provides a mechanism for connecting virtual 

cables between input and output points. This patching can 

occur between the physical inputs or outputs and DSP 

applications, or between DSP applications. 

Rather than provide individual volume controls within each 

DSP application, the Huron System Tools consolidate the 

volume controls in the IO Manager. This tool provides a 

volume control slider for each physical input or output. A 

radio-button control can be used to select the preamplifier gain 

for the input channels. When combined, these controls allow 

signal levels to be matched to the full-scale of the digital 

domain thus maximising the digital signal to noise ratio. 


Introduction 

Features 

The VRack Tool 

The “Virtual rack” or VRack application is the launch point for 

Lake Technology’s audio processing software. The VRack is 

designed to be analogous with a standard 19” audio rack system 

found in professional audio installations. It allows the user to 

construct a multitude of signal processing configurations, by 

adding virtual rack units to the graphical rack. 

The centralisation of controls within the VRack means that all 

software tools are accessible from a single interface. 

• Provides an interface to manage Lake Technology’s 

software tools. 

• Checks hardware configuration and allows the setting of 

channel names. 

• Provides a visual indicator of the total number of DSPs 

available in the system, and the current number initialised. 

• Allows the saving of the current workspace for subsequent 

recall. 


Operation 

Starting 

↓ To start the VRack tool 

Figure 8 — The VRack Interface 

THE SYSTEM TOOLS 

The VRack tool is started from the Windows 2000 Start Menu. 

1. From the Start Menu select Huron 3.2. 

2. Click on VRack. 

After the splash screen appears, the VRack will 

start. 

At the top of the VRack is the Rack Menu. This menu is used 

to select configurations, set various user options and exit the 

VRack. The menu is also used for putting new rack units into 

the VRack itself. 

Menu Item Contains 

File Options for saving, loading and naming VRack 

workspace configurations. 

Options Review the hardware configuration, and adjust the 

properties of the VRack. 

New Loads the virtual rack modules into the VRack. 

In the centre of the VRack are the Rack Spaces. When a DSP 

software or engineering application is launched, an icon 

representing the program will fill one of the spaces, these icons 

are known as Rack Panels. 

Each rack panel describes an application using a word and a 

symbol. These word/symbol associations are used throughout 

the DSP tool suites to allow quick visual identification of 

programs and their function. Each panel also has an Indicator 


Checking Configuration 

↓ To check configuration 


LED. This LED is off when the DSP program is hidden but 

running off-screen, and set to bright green when the DSP 

program window is displayed on the screen. 

There are an unlimited number of rack spaces in the VRack. If 

many DSP programs are launched the VRack display space may 

become full. Once the on screen portion of the VRack is full, 

all subsequently launched program panels are placed 

underneath those shown in the VRack. To access these panels 

use the Rack Space Scrollbar to bring the unseen portion of 

the VRack into view. 

The DSP Meter at the bottom of the VRack provides a visual 

indication of the number of DSPs initialised by the software. 

The meter describes what proportion of the DSP processors in 

the system are in use. To the right of this ratio is a graphical 

meter providing a proportional view of DSP usage. 

The VRack may be used to review what Lake hardware is 

installed in the system. 

From the VRack Menu. 

1. Select Options. 

2. Select Configure. 

This produces the following window: 

Figure 9 — The VRack Configuration Panel 

The right hand side of the window shows a summary of the 

hardware installed on the system. 


Configuration Functions 

Internal/External 

Sampling Rate 

Base Address 

No. Boards 

DSP Boards 

Processors 

IO Boards 

ADAT Boards 

IO Channels 

↓ To set the name of a channel 


The system sampling clock source can be selected from either 

the internal system clock, or an external clock derived from a 

digital input source. For an external clock to be selected the 

system must have the Huron ADAT card positioned in slot 0 or 

digital inputs on channels 15 and 16. No DSP resources can be 

in use while the switch is being made. 

The current sampling rate is displayed. If an external clock is 

selected, the clock source must be stable when the external 

selection is made. This frequency is assumed to never change 

after external is selected. 

The physical memory address in the PC where the Lake 

hardware resides. 

The total number of Lake boards installed in the system. 

The total number of DSP boards installed in the system. 

The total number of DSP processors installed in the system. 

The total number of Input/Output (I/O) boards installed in the 

system. 

The total number of Huron ADAT boards installed in the 

system. 

The total number of physical I/O channels in the system. 

The left side of the configuration window contains a listbox 

displaying the names of all the physical Input and Output 

channels. The first time VRack is run these names will be set to 

the factory defaults. The Scrollbar to the right of the listbox can 

be used to scroll through this list. 

The names of the channels may be set by the user. 

• Double click on the channel name which is to be changed. 

The following window appears: 

Figure 10 — Channel Name Setting 


Tip: Channel Names are saved along 

with workspaces (*.WSP) and will be 

restored when a workspace is 

reloaded. 

Configuration 

↓ To set VRack to always on top 

↓ To restore VRack’s normal - 

behaviour 

Launching Tools 

↓ To launch an application or 

DSP tool 

Note: It is possible to have more than 

one instance of a particular tool in the 

VRack at a time, (eg. the Convolver). 

If an attempt is made to run more 

than one copy of a program which 

does not allow multiple instances (eg. 

the Patch Bay), VRack will not load 

the program into the VRack. 


• Type a new name into the Name field. 

Pressing the OK button will save the new name. 

— or — 

Pressing Cancel will cancel the name change. 

Once a channel has been assigned a name, all software will 

show the new name of that channel. The appropriate naming of 

channels mapped to physical input and output devices makes 

the identification of specific channels or devices within DSP 

applications simpler. 

• After setting channel names, pressing the OK button in the 

Configure Window will keep any new channel names. 

Pressing Cancel will restore all channel names to their 

previous settings. 

The VRack can be configured to always be at the forefront of 

the desktop. 


1. Select the Options menu. 

2. Click Always on top. 

• Repeat the menu selections shown above. 

The VRack is used to start all applications or DSP tools. 

1. Select New from the Rack menu. 

2. Select the tool you require from the set of tool groups which 

appear in the drop down menu. 

The selected tool will be placed into the VRack interface, where 

it can be accessed by a single click. 


Displaying, Hiding and Terminating Tools 

↓ To display a DSP tool 

↓ To hide a DSP tool 

Note: Hiding the tool does not 

terminate the DSP tool and its 

functions 

↓ To terminate a DSP tool and 

remove it from the VRack 

Workspaces 

↓ To save a workspace 


When a DSP tool is first placed into the VRack, only its rack 

panel can be seen. In order to adjust the settings of the tool it 

needs to be displayed. 

• Click on the tool’s rack panel in VRack. 

The window representing the selected DSP tool will appear, and 

the Indicator LED on the VRack panel will change to bright 

green. This change indicates that the DSP tool is now 

displayed. 

Either: 

• Click on the rack panel of the tool you wish to hide 

— or — 

Click on the at the top right corner of the DSP tool. 

The rack panel’s Indicator LED will darken, and the application 

window will be hidden. 

1. Display the DSP tool by clicking on its rack panel. 

2. Select Close from the DSP tool’s system menu ( 

top right of the tool). 

The tool will terminate, relinquish its DSP resources and 

remove itself from the VRack. 

at the 

The VRack can be made to save the current set of DSP tools, 

the current Patch Bay settings, and the Input and Output settings 

in the rack space. This enables the VRack to be restarted with 

the same configuration as when it was saved. Additionally, the 

DSP tools will be restarted in their current state. These save 

files are known as workspace files (*.WSP). 


1. Select File. 

2. Select Save. 

This will save the workspace under the current workspace 

name. If no name has been given to the workspace, you will be 

prompted with a Save As… dialogue box, See Below. 

It is possible to save multiple workspace configurations by 

saving each workspace configuration under a new workspace 

name. 


↓ To save a workspace under a 

new name 

↓ To load a saved workspace 


From The VRack Menu. 

1. Select the File Menu. 

2. Select Save As… 

3. This activates the Save As… dialogue box (see Figure 11), 

which allows the creation of a name under which the 

workspace file will be saved. 

Figure 11 — The VRack Save As dialogue box 

4. In the File Name box enter the name you wish to save the 

workspace as. 

5. Click OK to save the workspace. 

The VRack configuration will be disassembled, saved and then 

restored from the file. 


1. Select the File Menu. 

2. Select Open. 

3. Select the required saved workspace from the listbox. 

4. Click OK to load the workspace. 

Alternatively, you can create a new icon with the saved 

workspace file in the command line. After creating this new 

icon you can automatically restore all of the DSP and Input and 

Output settings in the workspace, by Double clicking on the 

shortcut. 

The command line required to launch the VRack application 

and an associated workspace file is: 

C:\Huron32\VRack32.exe [drive and path to 

wsp file]\[name of wsp file].wsp 


Stopping VRack 

↓ To exit the VRack, and 

terminate all of the DSP tools 

Summary 


From the VRack Menu 

1. Select File. 

2. Select Exit. 

Each DSP tool will automatically be terminated and the VRack 

will close down. 

• The VRack is the core Lake Technology software 

application. It manages all of the DSP tools, controlling the 

starting, display and hiding of each one. 

• VRack provides a visual indication of all tools which are 

running and the amount of DSP processing power utilised 

and available in the current configuration. 

• VRack can save the current DSP tool configuration in a 

workspace file, allowing the VRack to start up and appear 

just as it was, when it was last saved. 


Introduction 

Features 

Operation 

Starting 

↓ To restart the Patch Bay 

The Patch Bay 

The Patch Bay is used to connect or patch audio between 

physical inputs and outputs and the Lake Technology 

applications. It provides a graphical representation of the 

connections between the DSP tools loaded into the VRack and 

the physical inputs and outputs present in the system. 

• Connects audio signals from physical inputs and outputs to 

DSP tools. 

• Connects audio signals between DSP tools. 

• Provides a graphical representation of all audio connections 

in the system. 

Figure 12 — The Patch Bay interface 

The Patch Bay is loaded automatically when VRack starts. 

However, provision exists for the Patch Bay to be reloaded 

from the VRack should it terminate due to a system error. 


Patching 

↓ To connect a pair of patch 

points 

↓ To connect multiple patch 

points, in series 



1. Select the New Menu. 

2. Select the Engineering Tools Submenu. 

3. Select the Patch Bay icon. 

The Patch Bay is divided into smaller areas called patchsets. 

Each patchset represents all the inputs and outputs; of a 

particular DSP tool, the set of physical inputs or, the set of 

physical outputs. 

Within each patchset there exists a number of patch points. 

Each patch point represents either an input or an output and can 

be connected via a graphical patch cable to another patch point. 

Patch points are the virtual equivalents of the XLR connectors 

found on audio hardware. 

Each DSP tool patchset may be assigned a name from the Patch 

Bay. This name will appear in the DSP tool name section of 

the display. 

Scroll pins on the physical input and output patchsets allow the 

pinning of the patchset. When a patchset is pinned it will not 

scroll when the Patch Bay’s scrollbar is used. 

The connection of audio between two patch points is done by 

selecting pairs of patch points. 

1. Click on the first patch point. 

2. The patch point will turn red signifying it has been selected. 

3. Click on the second patch point. 

If compatible, the two patch points will be connected by a 

patch cable. 

1. Click on a set of patch points. 

The patch points will stay red indicating they are all selected. 

2. Click on the first point in the series that you wish to connect 

the original patch points to. 

The patcher will connect the patch points in series. 

Figure 13 — An example of multiple patching 


Valid Patching 

↓ To de-select a selected patch 

point 


A valid patch is a patch made between two differing patch point 

types, which can be recognised in the patcher: 

white) 

Legal Patches 

From 

To 

Physical Input DSP tool input 

DSP tool output DSP tool input 

DSP tool output Physical output 

Physical input or DSP tool output (graphical XLR 'pins' are 

Physical output or DSP tool input (graphical XLR 'holes' 

are black) 

When a valid patch has been made, a patch cable is drawn 

between the two patch points indicating that they have been 

patched together, and both patch points will become black in 

colour. 

• Click on the red selected patch point a second time. 

The patch point will return to its usual colour and a new 

selection may be made. 

Only the following patches are legal. Attempts to patch illegal 

patch points together results in an audible beep being sounded 

from the system and the patch request being ignored. 

In addition, only some patch points may have more than one 

patch cable attached: 

Type of Patch Point Allows 

DSP input Only one cable attached 

Physical output Only one cable attached 

DSP output Multiple cables attached 

Physical input Multiple cables attached 

If a new DSP tool is started, the Patch Bay will automatically 

add a new patchset. Conversely, if a DSP tool is stopped, the 

patchset will disappear. 

Some DSP tools also change their number of inputs and outputs 

during the course of their execution. The Patch Bay notices any 

changes in DSP tool channels and adjusts itself accordingly. 

Sometimes patchpoints will appear as though they are 

connected (black) but will not have a patch cable connecting 

them to anything. These points are “hidden” and are in use 

when the Patch Bay is in an alternate state (this effect can be 

seen when switching between Convolver programs that have 

different numbers of filters). It is possible to deselect and 


Tip: Even if two patch points are 

connected, you may need to adjust 

gain settings and other parameters of 

the DSP tool before any sound is 

actually heard. 

Unpatching 

↓ To remove a patch cable from a 

patch point 

Note: If there are multiple patch 

cables attached to the patch point that 

is double clicked then all patch cables 

attached to that patch point are 

unpatched and removed. 

Naming DSP Tools 

↓ To name a DSP tool 

Note: The name in the Name section of 

a DSP tool will not save multiple 

words separated by spaces. To 

separate words use an underscore 

or 

hyphen. 

Scrolling 


repatch these points, however this removes the alternate state 

settings. 

• Double click on one of the patch points of the patch cable to 

be removed. The patch point should not have been 

previously selected (ie. not red but black) before double 

clicking. 

The Patch Bay allows the user to assign a name to each of the 

DSP tools currently running. This function can be useful when 

more than one of the same DSP tool is loaded into the VRack. 

1. Type the new 

name into the 

DSP tool name 

area. 

2. Press Enter. 

Figure 14 — DSP tool naming 

The caption in the DSP tool name area will change. 

When there are many patchsets or physical inputs and outputs, 

then the Patch Bay window may be scrolled up and down to 

bring hidden patchsets into view. Scrolling the window does 

not de-select a previously selected patch point. Thus, it is 


↓ To pin a patchset 

↓ To unpin a patchset 

Summary 


possible to patch a patch point located at the top of the window 

to one at the bottom of the window by selecting the first patch 

point, scrolling the window, and then selecting the second patch 

point. Scrolling of the window is done using the scrollbar at the 

right hand side of the Patch Bay. 

The physical input and output patchsets also have a feature 

where they may be pinned to the window. When a patchset is 

pinned, it does not scroll when the Patch Bay window is 

scrolled. 

• Click on the pin symbol or 

the arrows next to the pin. 

The symbol will change to the pinned symbol, and the patchset 

will be pinned. 

• Click on the pinned symbol or the arrows next to the pinned 

symbol. 

The symbol will change to the pin symbol and the patchset will 

be unpinned. 

• The Patch Bay allows the audio patching of DSP tools, 

physical inputs and physical outputs, by graphically 

selecting two patch points with the mouse. 

• Patches are shown on the Patch Bay as lines between patch 

points, these lines are known as patch cables. 

• The Patch Bay updates itself as DSP tools are started and 

stopped. 

• DSP tools may be assigned a name using the Patch Bay. 


Introduction 

Features 

The IO Manager 

The IO Manager is used to adjust the gain levels of the system’s 

physical analogue inputs and outputs. The IO Manager also 

provides status information for the systems digital inputs and 

outputs. 

• Allows volume adjustments of analogue inputs and outputs. 

• Allows the setting of the gain for analogue inputs. 

• Provides a graphical overview of all physical analogue 

inputs and outputs, their associated volumes and input gain 

levels. 

• Provides access to the status of digital inputs. 

• Allows the mode selection of digital outputs. 


Operation 

Starting 

↓ To access the IO Manager 

Figure 15 — The IO Manager interface 


The IO Manager is started automatically when VRack is 

launched. 

• Click on the IO Manager icon from the VRack. 

The IO Manager is divided into two areas: 

Physical outputs are placed at the top of the window 

Physical inputs are placed at the bottom. 

Each slider represents a physical Input or Output installed 

within the system. 

Each slider lists the physical channel number that it controls 

and the current volume setting in dB. Additionally, analogue 

inputs have pre-amp gain levels which may applied. 


Adjusting Volumes 

↓ To adjust the volume of a given 

channel 

Adjusting Input Level Gain 

↓ To adjust the input gain level of 

an input channel 

Note: The volume and input level 

settings are effective throughout the 

system, and will be saved when the 

current workspace is saved. 

Muting an Output 

↓ To mute an output channel 

↓ To unmute an output channel 

↓ To mute all outputs 

↓ To unmute all outputs 

Either 

Figure 16 — Controls for adjusting volume 


• Click the on the Volume slider bar to select the desired 

volume level. 

— or — 

Drag the Volume slider bar to the desired level. 

— or — 

Type in a volume level from -30 to +30 dB. 

• Click on the desired input gain level. 

The current gain level will be set to the level indicated by the 

graphical switch. 

• Click on the Mute box above the channel to mute. 

• Click on the Mute box above the channel which is muted. 

• Click Mute All. 

• Click the previously selected Mute All. 


Viewing Digital Input Status 

↓ To view the status of a digital 

input 


1. Check the colour of the digital input status LED. It will be 

green when no input errors are detected, yellow when nonfatal 

input errors are detected and red when fatal errors are 

detected. 

2. Click the LED to activate a status window. 

Figure 17 — Digital input status 

Lake Technology’s software reports nine error conditions that 

may occur on digital input channels. 

Error Reported Description 

No Error The digital input system is not producing any errors. 

Validity bit high The input is working, but audio input is not being 

generated (eg. Pause or Stop has been pressed on the 

input device). 

Confidence lag The input signal has a high rate of noise, or is not of 

good quality. 

Slipped Sample An uncompensated difference in the sampling rates of 

the input device and the DSP machine are causing loss 

or duplication of samples. 

CRC error Professional inputs only - a CRC error has been 

detected in the channel status signal. 

Parity Error Parity checking has returned an error. 

Bi-Phase coding A signal is being produced on the input, but it does not 

have the correct format. 

No PLL Lock The line is receiving no signal whatsoever. 

Unknown error Error condition unknown. 

Setting Digital Output Format 

Some digital outputs can be encoded by Lake equipment in 

either a professional or consumer format. Each format uses 

different codes for the channel status bits placed in the digital 

output stream. The digital output from the Lake system should 

thus be set in accordance to the format of the equipment 

accepting the digital output. 


↓ To set a digital output to 

professional or consumer output 

Grouping Volume Sliders 

↓ To group volume sliders 

together 

Showing Channels in a Group 

↓ To determine if a channel is part 

of a group 


• Click the Pro or Con digital output toggle to select either 

Professional format or Consumer format. 

The toggle will move to indicate your selection. 

Figure 18 – The IO Manager channel manipulation features 

1. Click the Channel Number Boxes of all the channels to be 

grouped. As each Channel Number box is clicked it will 

highlight. 

2. Click the Group box. 

The Grouped Sliders will all be set to the volume level of the 

lowest numbered slider in the group, with inputs having 

precedent over outputs. 

When grouped, changes to the volume of one slider will affect 

all sliders in the group. 

1. Click the Channel Number Box of the channel whose 

group you wish to determine. The box will highlight. 

2. Click the Show Box. 

All channels in the same group as the highlighted Channel 

Number Box will be highlighted. 


Removing Channels from a Group 

↓ To remove channels from a 

group 

Splitting Channels 

↓ To split volume controls to a 

new window 

Split Windows 

↓ To change the name of a split 

window 


1. Click the Channel Number Box of the channels to remove 

from a group. 

2. Click the Ungroup Box. 

Input and output sliders can be cloned from the IO Manager 

panel into their own window. This splitting of channels allows 

the user to focus on controlling the gain of selected channels. 

1. Click the Channel Number Box of the channels to be split. 

Each click should highlight a channel. 

2. Click the Split Button. All of the channels in the selected 

group will re-appear in their own window. 

Figure 19 — Split volume control window 

Split channels appear in a separate window. They behave in the 

same way as normal volume adjustment channels, with the 

exception that you cannot group from, or split from, a 

previously split window. 

1. Click in the title box of the split window. 

2. Type a new name for the split window. 

3. Press Enter. 


↓ To close a split window 

Summary 


• Click the system menu box in the top left hand corner of the 

window. 

• The IO Manager is used for setting input and output gain 

levels, viewing digital input status, and setting digital output 

format. 

• The current values of channel volumes can be read 

graphically and textually in the IO Manager. 


Introduction to the Simulation Tools 

The simulation tools are comprised of a number of separate 

VRack applications that provide simulation of threedimensional 

acoustic spaces over an array of speakers or a set 

of headphones. 

Binscape: Which provides facilities for simulating virtual 

acoustic spaces with multiple moving sound sources. Binscape 

uses binaural processing and optimised panning algorithms to 

provide high-end positional audio over headphones. 

Space Array: Which provides facilities for simulating multiple 

moving sound sources and three-dimensional virtual acoustic 

spaces. Featuring an optimised panning algorithm for speaker 

arrays, Space Array has an enlarged sweet spot and is suitable 

for personal or group presentations. 

AniScape: Which provides facilities for simulating threedimensional 

virtual sound spaces, where a single user can hear 

multiple moving sources through a speaker array or 

headphones. Aniscape is made up of a number of individual 

simulation tools, including Sound Field Filter, Speaker 

Decoder, and Binaural Decoder. Binscape and Space Array 

have superseded Aniscape for most simulation applications. 

MultiScape: Which provides facilities for simulating threedimensional 

virtual acoustic spaces, where multiple users can 

hear multiple moving sources by using a speaker array or 

headphones. Binscape and Space Array have superseded 

Multiscape for most simulation applications. 

HeadScape: Which provides facilities for using long, low 

latency FIR filters in combination with a head tracking device. 

This combination allows HeadScape to produce very accurate 

three-dimensional sound for high-end acoustic simulations. 

Sonic Animator: Which provides a graphical interface and 

panning tool for manipulating audio sound source location. 

Lake Technology provides a selection of Other simulation tools 

which integrate with the above tools and other Lake 

Technology VRack applications. These other tools include: 

Locator: Designed for testing and configuration verification, 

the Locator provides a visual interface to allow the positioning 

and movement of sound objects on a two dimensional plane. 

WavePlayer-2: An enhanced version of WavePlayer for play 

back of long duration Wave files (*.WAV) with zero Huron 

DSP usage. 

WavePlayer: Designed to play back Wave files (*.WAV) 

encoded in the *.SIM file format through the Huron system. 

Tracker: Allows the Huron to interface with a threedimensional 

position tracking system such as the Polhemus® 


THE SIMULATION TOOLS 

InsideTrak, Fasttrak, Ascension Flock Of Birds, Intersense IS- 

300, IS-600, IS-900, InterTrax (serial) and InterTrax-2 (USB). 

Please contact Lake Technology for further information about 

Huron tracker interfaces. 

Socket: Enables the remote control of a Huron system via 

TCP/IP control packets from a remote devide. The SNAP 

protocol is used for communication with Socket. 

MakeRoom: A windows application designed for the statistical 

generation of virtual rooms. These rooms are designed for use 

with the Simulation Tools or the Engineering Tools 

applications. 

In addition to VRack applications, Lake Technology provides 

the following API for controlling Simulation Tools VRack 

applications. 

SNAP: An API for controlling the Huron from an external 

application. The external application can be located on the 

Huron or another computer on the local network. 


An overview of the Simulation Tools 

Lake Technology’s Simulation Tools provide a wide range of 

options for configuring and creating aural simulations. This 

multitude of potential configurations provides not only a wealth 

of possibilities for aural simulations but introduces a level of 

complexity to the running of such simulations. 

In order to reduce possible confusion this subsection seeks to 

provide an overview of the relationship between the elements 

which comprise a simulation and addresses the most important 

aspects of setting up a Lake Technology sound simulation. 

The following terms are used throughout the simulation tools 

documentation and are important to know in order to 

understand the functioning of the Simulation Tools. 

• Real world: This is it, you are living in it. 

• Soundscape: The virtual audio simulation generated by a 

simulation application. The Soundscape can be thought of 

as a virtual audio world. 

• Simulation application: An application which takes audio 

and other inputs and processes them to generate a 

Soundscape. 

• Listener: The real world person who listens to an audio 

simulation via headphones or a speaker array. 

• Listener object: The virtual object that represents the 

listener inside the Soundscape. 

• Sound source: The real world sound source (CD player, 

WavePlayer application etc.) that is used as an audio input 

into the simulation application. 

• Sound object: The virtual object that represents the sound 

source in the Soundscape during a simulation. 

• Position: The real world position of the listener or tracking 

device. 

• Location: The virtual location of an object (listener or 

sound) within the Soundscape. 

• Location channel: The messaging channel over which 

information relating the location of virtual objects is passed. 

Location channel numbers must be in the range 1 to 255, as 

channel 0 is reserved in most applications. 

• Rotation: The real world orientation of the listener or 

tracking device. 

• Angle: The virtual orientation of a sound or listener object 

in a Soundscape. 



• Angle channel: The messaging channel over which 

information relating the orientation of a virtual sound or a 

listener object is passed. Location channel numbers must be 

in the range 1 to 255, as channel 0 is reserved in most 

applications. 

• Tracking device: A device used for obtaining and sending 

positional and rotational data. This data is then transformed 

(by the Tracker application) into location and angle data for 

use as an input to a simulation application. 

• Speaker array: A real world array of speakers designed for 

the playback of audio simulations. 

Additional terms can be found in the Glossary subsection of the 

Huron Technical Specifications of this manual. 

In general terms an aural simulation is composed of a 

processing application — the simulation application — and a 

series of inputs to this application. The diagram below (Figure 

20) illustrates a generic simulation tool and lists the sources of 

inputs and outputs of the system. 

Figure 20 — A schematic of a generic Simulation application 

The simulation application is the core of an aural simulation. It 

takes the sound sources, location information, angle 

information, and room effects as inputs, and processes all of this 

information in real time to provide the outputs of the simulation 

either as B-format encoded signals (as in Sound Field Filter) or 

in a form ready to be output to a speaker array or headphones 

(Binscape, Space Array , MultiScape, HeadScape or a 

combination of Sound Field Filter and one of the AniScape 

Decoder tools). 

An important concept shown in this diagram (Figure 20) is the 

association of the manipulable virtual sound objects and the 

sound sources they represent. The sources and objects are 

associated using Location and Angle channels. While the real 

world sound sources remain stationary (locked into equipment 

racks etc.) the sound objects may be moved around the 



Soundscape through Angle and Location messages sent from 

applications such as Socket, Locator or Sonic Animator. 

The following diagram (Figure 21) illustrates the interaction 

between elements of a typical simulation. This simulation uses 

MultiScape as the simulation application, and Locator for the 

loading of room responses and the movement of sound objects 

(details for each of these applications are found in the 

appropriate subsections of the simulation tools). 

Figure 21 — The interaction of several simulation tools to create an aural simulation 

Coordinate Systems 

Lake’s simulation tools use a right-handed coordinate system to 

specify the location of sound sources. When the user is 

stationary and located at the origin of the system, the Y-axis 

extends in front of the user, the X-axis is to the user’s right, and 

the Z-axis extends upwards, as shown in Figure 22. 


Z 


Figure 22 — The Lake coordinate system 

In some tools, a B-format audio stream is used. B-format audio 

relies on the theory of Ambisonics, whose coordinate system 

differs from the Lake system. In Ambisonics, the X-axis is in 

front the user, the Y-axis extends to the left, and the Z-axis 

extends upwards, as shown in Figure 23. 

Y 

Z 

Figure 23 — The Ambisonic coordinate system 

These coordinate systems specify the location of sound sources 

in a 3-dimensional coordinate space. As well as location, sound 

sources are also described by an orientation or angle, which 

specifies the direction the sound will travel from the source. 

Lake uses the azimuth, elevation, and roll paradigm to describe 

the orientation of sound sources, as shown in Figure 24. 

HURON TECHNICAL MANUAL PAGE 70 

Y 

X 

X

Calibrating Speaker Arrays 


Azimuth 

Elevation 

Roll 

Figure 24 — Positive azimuth, elevation, and roll of a sound 

source 

Given the forward-facing vector of a sound source, azimuth is 

defined as a rotation of the vector about the X-Y plane; 

elevation is a rotation towards the Z-axis; and roll is a rotation 

of the source about the direction of the vector itself. 

When rendering a simulation to a speaker array, it is desirable 

to calibrate the amplifier gains so that each speaker contributes 

equally to the sound pressure level at the centre of the array. 

This will eliminate unbalances in the array so that panning is 

smooth. The technique described below is valid only for arrays 

where the speakers are equidistant from the centre of the array. 

A sound pressure level meter can be used to measure the output 

power of a given amplifier / speaker configuration. Feed a 

constant input source such as a 1kHz sine wave into one 

speaker at a time and measure the resultant on-axis output 

power at a fixed distance (say 1 metre) from the speaker. 

Adjust the amplifier gains so that each speaker has the same 

volume with the given input source. 


Features 

Operation 

Object list 

Add source 

object 

Edit object 

Delete source object 

BinScape 

BinScape is a 3D positional-audio simulator based on binaural 

technology. It can simultaneously render up to 32 sound 

sources to a single listener position. BinScape uses only 3 

DSPs regardless of the number of sound sources and provides 

for headtracked listening via SNAP commands. 

BinScape can load user definable head and room filters for 

engineering research applications. 

• Fixed usage of only 3 DSPs for 3D rendering of 1 to 32 

source objects to a single listener. 

• Simulation of 3 rooms with varied acoustical damping and 

size properties. 

• Head tracking is possible by setting channels for BinScape 

to receive listener position and orientation messages from a 

tracker tool. 

• User supplied head and room filters can be loaded. 

• Can be controlled remotely via SNAP commands and the 

Socket application. 

Reset object position 

Select head filter set 

Figure 25 — Customised BinScape main interface 

Select 

Room 

filter 


Starting 

↓ To access BinScape 

Editing the Listener Object (receiver) 

↓ To edit the Listener (receiver) 

Object 


BinScape must be loaded into the VRack through the 

New/Simulation Tools submenu (For loading instructions, see 

the System Tools section of this manual). Once loaded, 

BinScape will appear in the VRack, and its associated patchset 

will appear in the Patch Bay. 

• Click on the BinScape panel in the VRack. 

The main BinScape window will appear (see Figure 34). 

BinScape provides outputs for a single headphone listener. 

1. Select the Listener Object in the BinScape object list. 

Click Edit. 

—or — 

Double-click the Listener Object. 

The following dialogue box will appear: 

Object name 

Primary 

location 

channel 

Primary 

orientation 

channel 

Secondary 

location 

channel 

Secondary 

orientation 

channel 

Accept changes Cancel changes 

Figure 26 — BinScape Listener Object edit dialogue 

2. Type a name for the Listener Object in the Name box. 

3. Set the Location Channel Number in the Primary 

Location channel box. This Channel number specifies the 

channel on which the object will receive SNAP or Locator 

messages indicating its current (X,Y,Z) position. It is not 

necessary to set a secondary Location Channel Number for a 

Listener Object. 

4. Set the Secondary Location Channel Number if the 

listener is to be moved relative to another object. 


For example: if a Source Object 

(channel 2) needs to be moved 

independently but relative to another 

Source Object (channel 1) its location 

should be specified as Location: 1+2. 

Thus, if the object using location 

Channel 1 moves the object on 

Channel 2 will move with it. 

For example: An elephant (whose 

facing is specified as angle channel 

number 1) has a trunk (whose facing 

is specified as channel number 2), the 

elephant’s trunk may rotate 

independently yet maintain a 

relationship to the facing of the 

elephant, thus the trunk’s angle 

channel number should be defined as 

1 + 2. 

Note: The first two elements in the 

BinScape object list must always be 

the Listener Object (receiver) and a 

Source Object. As it is necessary to 

have at least these two elements to run 

a simulation. You cannot delete the 

listener or a sole Source Object. 

Adding and Removing a Source Object 

↓ To add a new Source Object 


5. Set the Angle Channel Number in the Primary 

Orientation channel box. This Angle Channel Number 

specifies the channel over which the orientation (facing) of 

the listener is transmitted. In the case of the Listener Object, 

it represents the channel over which headtracking or facing 

information will be transmitted. 

6. Set the Secondary Angle Channel Number if the listener is 

to be rotated relative to another object. 

7. Click OK to accept the changes to the Listener Object. 

—or — 

Click Cancel to cancel the changes to the Listener Object. 

1. Click the Add button in the main BinScape window. 



Source name 

Primary 

location 

channel 

Primary 

orientation 

channel 

Directivity Factor Name 

d = 0.000 Omni 

d = 0.500 Cardioid 

d = 0.625 Super Cardioid 

d = 0.750 Hyper Cardioid 

d = 1.000 Figure 8 or Dipole 

↓ To remove a Source Object 

from the BinScape object list 


Secondary 

location channel 

Secondary 

orientation channel 

Directivity factor 

Add sound object Cancel add object 

2. Type a name for the Source Object in the Name box. 

3. Set the Location Channel Number in the primary Location 

box. This Channel number specifies the channel on which 

the object will receive SNAP or Locator messages indicating 

its current (X,Y,Z) position. To ensure that the source can 

be moved independently of other sources this Channel 

Number should be unique. 

4. Set the Secondary Location Channel Number if the source 

is to be moved relative to another object as described for the 

Listener object above. 

5. Set the Angle Channel Number in the Primary Orientation 

channel box. This Angle Channel Number specifies the 

channel over which the orientation (facing) of the source is 

transmitted. 

6. Set the Secondary Angle Channel Number if the source is 


7. Set the Dir Fact (Directivity Factor) of the Source Object. 

The Directivity Factor determines the shape of the sound 

field each source will produce. Some common directivity 

patterns are given below: 

8. Click OK to add the Source Object to the BinScape object 

list. 

—or — 

Click Cancel to cancel the addition of the Source Object. 

From the main BinScape window 

1. Select the Source Object to remove by clicking on the 

Source Object in the BinScape object list. 

2. Click Delete. 

The Source Object will be removed from the BinScape object 

list. 


Editing a BinScape Object 

↓ To edit the details of Source 

Objects 

↓ To reset the position of a 

Listener (receiver) or Source 

Object 

Room Switching 

↓ To switch rooms 

Custom Room Filter Specifications 


1. Select the Source Object to edit from the BinScape object 

list. The source object will be highlighted. Click Edit. 

Alternatively, double-click the Source Object in the 

BinScape object list. 

2. Edit Name, Location channels, Angle channels and 

Directivity Factors as required. 

3. Click OK to accept the changes, or click Cancel to ignore 

the changes. 

1. Select the Sound or Listener Object to edit by clicking on the 

Source Object in the BinScape object list. 

2. Click Reset. 

The sound or Listener Object location will be set to the origin 

point at 0.00, 0.00, 0.00. 

BinScape allows the simulated room to be switched smoothly 

between three options representing spaces of varied acoustical 

damping and size properties. 

• Switch the current room filter using the Room listbox on the 

main dialog or by sending the SNAPSetRoom command 

while the Socket application is running. 

BinScape allows the user to load a new binaural room filter that 

replaces the first standard room filter set. The user can then 

switch between the remaining standard rooms (Two and Three) 

and the new Custom room. 

1. Load a new binaural room filter set by selecting "Load 

Custom…" from the Room listbox and following the 

prompts to load the Left and Right filters. 

2. The custom filter will replace Room One on the DSP. You 

can now switch between the Custom room and standard 

rooms Two and Three. 

3. A valid custom Room filter set consists of two .sim files, 

each 6144 samples long, where the first sample of each 

filter occurs at t=384*Ts, Ts=sample period. Ie, you can 

make a valid Room filter file by taking a complete binaural 

room impulse response for each ear, discarding the first 384 

samples and truncating the result at 6144 samples long. 


Custom Head Filter Specifications 

Creating Valid Head and Room Filter Files Using Matlab 

Summary 


BinScape allows the user to load a new set of binaural head 

filters in addition to the existing head filters. 

Head filters provide the directionality of BinScape 3D rendered 

sound sources. The user can then switch between the standard 

head filters and their custom set of head filters. 

1. Load a new set of head filters by selecting "Load Custom…" 

and following the prompts to load the Centre, Front 

Common, Rear Common, Front Difference and Rear 

Difference filters. 

2. You can now switch between the Standard head filters and 

the Custom head filters by selecting from the listbox. You 

can also load a new set of custom head filters over a 

previously loaded custom set as above. 

3. A valid set of custom head filters consists of five .sim files, 

each 128 samples long. The first sample of all head filters 

occurs at t=0. 

4. The meaning of the five head filters is as follows: 

5. Centre: average of the HRIRs to both left and right ears for 

a source at 0 degrees - ie directly in-front of the listener. 

6. Front Common: common-mode portion of near and far ear 

HRIRs to a source at 45 degrees to the listener direction. 

7. Front Difference: difference-mode portion of the near and 

far ear HRIRs to a source at 45 degrees to the listener 

direction. 

8. Rear Common: common-mode portion of near and far ear 

HRIRs to a source at 135 degrees to the listener direction. 

9. Rear Difference: difference-mode portion of near and far 

ear HRIRs to a source at 135 degrees to the listener 

direction. 

For an example of how to make valid and proper Head and 

Room filter .sim files for BinScape Custom, see the Matlab 

script file makecustomfilters.m, supplied with BinScape 

Custom. 

• BinScape performs 3D positional-audio of up to 32 source 

objects to a single listener position. 

• BinScape has filters to simulate 3 rooms of different acoustic 

properties. Headtracking is also possible with additional 

tracker tool hardware. 


Features 

Operation 

The Space Array 

Space Array is a tool designed for animation and real-time 

acoustic rendering of sounds across three-dimensional speaker 

arrays. Up to 32 discrete sound sources may be animated 

within a soundscape. In addition to accurately rendering given 

sound sources, the array will also add room simulation based on 

pre-calculated reverberations. 

• Up to 32 sound sources animated over a maximum of 50 

speakers. 

• Large sweet spot within the speaker array, so the simulation 

will reach the entire audience. 

• Dynamically calculated speaker response of individual 

components within the speaker array makes physical speaker 

placement less critical to the overall sound reproduction 

quality. 

• Accepts input signals in B-format. 

The Space Array renders sound sources arriving through the 

patchbay inputs to speakers connected to patchbay outputs. It 

uses Lake proprietary algorithms to ensure that the sound is 

panned as smoothly as possible when sources move around the 

array. 


Starting 

↓ To access the Space Array 

The Space Array Object List 

Figure 27 – The Space Array interface 


The Space Array must be loaded into the VRack from the 

New/Simulation Tools submenu (for loading instructions see 

the System Tools section of this manual). Once loaded the 

Space Array panel will appear in the VRack and the Space 

Array panel will appear in the Patch Bay. 

• Click the Space Array panel in the VRack. 

The Space Array interface will appear (Figure 27). 

Space Array allows a user to define virtual objects and move 

them around within a soundscape. The Space Array Object List 

specifies all objects found in a soundscape and how they are 

moved around within it. 

In Space Array, there are two types of object: the listener 

object, and the sound source. There is only one listener object, 

which represents all listeners in the soundscape. The listener 

must be the first object in the object list. Every subsequent 

object in the list represents a sound source. 

In Space Array, a channel number is associated with each object 

in the list. The channel number specifies a channel on which 

the object will receive position and angle messages. 


Adding an Object to the Space Array Object List 

↓ To add an object to the Space 

Array list 

For example: if a sound object 

(Channel 2) needs to be moved 


sound object (Channel 1) its location 






facing is specified as Angle Channel 

Number 1) has a trunk (whose facing 

is specified as Channel Number 2), the 




elephant, thus the trunk’s Angle 

Channel Number should be defined as 

1 + 2. 

Editing an Object 

↓ To edit the details of Sound 

objects 


1. Click Add. A dialogue box will appear (this dialogue box 

appears the same for both the listener and sound source 

objects). 

2. Type a name for the object in the Name box. 



the object will receive SNAP or Locator messages indicating 




4. Set the Secondary Location Channel Number if the object 

is to be moved relative to another object. 



channel over which the orientation (facing) of the object is 

transmitted. 

6. Set the Secondary Angle Channel Number if the object is 


7. Click OK to add the object to the Space Array object list. 

8. Click Cancel to cancel the addition of the object. 

1. Select the Sound object to edit by clicking on the Sound 

object in Space Array list. 

2. Edit Name, Location channels, and Angle channels as 

required. 


Removing an Object 

↓ To remove the Sound object 

from the Space Array list 

Adjusting Source Characteristics 

Diffusivity 

Diffusivity Balance 

Reverb Balance 

3. Click OK to accept the changes. 

—or — 

Click Cancel to ignore the changes. 


From the main Space Array window 

1. Select the Sound object to remove by clicking on the Sound 

object in the Space Array list. 


The Sound object will be removed from the Space Array list. 

It is possible to tailor the playback characteristics of each sound 

source object in the Space Array object list. These source 

characteristics adjust the directionality, smearing, reverberation 

and behaviour of the sounds. 

The diffusivity level of the sound source determines the 

smearing effect on the sound by blurring the apparent 

directionality of the sound. The higher the smear the more 

diffuse the sound, the lower the smear the more precise the 

sounds location. 

The diffusivity balance determines where sound appears to 

radiate from and how localised the sound is. For example, A 

high smear, high balance gives a localised sound with lots of 

smear, while a high smear, low balance means sound appears to 

come from around the location of the object, rather than directly 

from it. 

Alters the amount of audio passed into the room reverberation 

filters used in the space array. The higher the value the longer 

the reverberation period. 


Near Field 

Adding B-format Input 

Setting Minimum Source Distance 

Defining the Speaker Array 

↓ To define the speaker positions 


The near field determines what happens when sound gets within 

the minimum source distance. If the near field is on flip and the 

sound source enters the minimum sound distance, the sounds 

will flip from one speaker to the next, while when near field is 

set at omni, the sound will appear to come from all of the 

speakers equally. 

The Space Array comes with an input for a B-format sound 

source. Such sound sources have 4 outputs which contain 

sound signals for x,y,z, and omni directions. This input can be 

used for interactive sounds within the soundscape. 

The minimum source distance represents the virtual distance at 

which sounds will be played back at unity gain. The manner in 

which objects within the minimum source distance are played 

back is altered by the sound object near field setting. 

Figure out how many speakers are in the array. Add each 

speaker and adjust locations of the speakers from an arbitrary 

origin point found at (0,0,0). The listener is assumed to be at 

0,0,0. If the listener is located elsewhere, enter the location of 

the listener. 

• Click Speakers. 

A new dialogue box will appear, allowing the array to be 

defined (Figure 28). 


Placing the Observer 

↓ To adjust the observers location 

Adding speakers 

↓ To add speakers to the array list 

Placing speakers 

↓ To place a speaker or edit its 

location 

Figure 28 – The Space Array Speakers dialog 


The observer’s location represents the physical position of the 

observers in the speaker array ie. the origin point for all the 

speaker activations. By default the observer is located at 0,0,0 

in the centre of the array, and the values given for the speakers 

are given in terms of that origin point in metres. 

1. Select the observer X field type new number. 

2. Select the observer Y field and type a new number. 

3. Select the observer Z field and type a new number. 

1. On the array list click Add. 

2. This will add a new speaker to the speaker list. 

1. Select the X location. 

2. Change the value. 

3. Select the Y location. 


5. Select the Z location. 



Inserting speakers 

↓ To insert a speaker into the list 

Deleting speakers 

↓ To delete a speaker from the list 

Pre-calculating Speaker Activations 

↓ To pre-calculate the speaker 

activations 

↓ To save speaker activations 

↓ To load a set of pre-calculated 

speaker activations 


1. Click Insert to add a speaker into the currently selected 

speaker location. 

2. Once added, select and edit the location values. 

1. Select the speaker to delete. 

2. Click Delete and the speaker will be removed from the list. 

Once a Speaker array has been defined it, needs the speaker 

activations to be pre-calculated. Calculating the speaker 

activations may require some time, however, once complete it 

provides access to the real time panning of sources across the 

speaker array. 

Once speakers have been set up, the speaker activations can be 

pre-calculated. 

• Click Calculate to calculate the speaker activations. 

A Cancel button window will appear to enable the calculation 

process to be cancelled if cancelled the speaker activations will 

not be calculated. The progress of the calculation may be 

checked in the status bar which progresses from left to right as 

the calculation continues. 

1. Click Save to save speaker activations. 

2. A Save File dialogue box will appear. 

3. Select or type a new filename and the pre-calculations will 

be saved in a speaker weight file with the *.3dw extension. 

1. Click Load to load previously calculated speaker 

activations. 

2. A dialogue box will appear. 

3. Select a *.3dw file. 

4. Click OK. 


Adding Room Reverberation 

↓ To add room reverberation 

↓ To remove room reverberation 

Moving Sound Objects 

DSP Usage 


Space array allows you to add room reverberations to the 

soundscapes generated by the space array simulation tool. 

These reverberations can be singular omni reverbs or can be 

based around B-format responses of real room spaces, which 

can be generated either though a simulation application like 

Catt-Acoustics or VRoom or MakeRoom, or recorded using 

Lake Technology’s Measure application. 

Either mono or B-format reverbs may be added to the 

simulation. B-format reverbs require more DSPs but allow the 

reverb to be accurately panned to the speakers. Mono reverbs 

render the reverberant tail equally to all speakers. 

If mono reverb is desired: 

1. Click the OMNI button to add the mono reverb SIM file. 

2. Find the Omni sim file (usually denoted by a w prefix). 

3. Click OK. 

If a B-format reverb is desired: 

1. Uncheck the Mono checkbox. 

2. Click the OMNI button to add the mono reverb SIM file. 

3. Find the Omni sim file (usually denoted by a w prefix). 

4. Click the X button to add the X plane SIM file. 

5. Find the X sim file (usually denoted by a X prefix). 

6. Click the Y button to add the X plane SIM file. 

7. Find the Y sim file (usually denoted by a Y prefix). 

8. Click the Z button to add the X plane SIM file. 

9. Find the Z sim file (usually denoted by a Z prefix). 

• Click the Clear button. 

Although sound objects can be moved in real time using a 

tracking device, Locator, or by a remote computer via the 

SNAP protocol. The Sonic Animator has been specifically 

designed for creating three-dimensional animated soundscapes 

for recording and playback. 

The number of DSPs used by the Space Array is dependent on 

three variables: the number of inputs (including the B-format 

inputs); the number of outputs (speakers); and the type of room 

reverberation used. The number of DSPs used can be 

characterised by the following formula. 

( ceil( 

i/ 

18) 

) * ( ceil( 

o 12) 

) 

n = 

/ 


Summary 

25 

20 

15 

Number of inputs 

Space Array DSP Usage 

0 

5 


Where: 

n is the total number of DSPs required. 

i is the total number of inputs. 

o is the total number of outputs. 

A graph illustrating the number of DSPs used according to the 

number of inputs and outputs is given below. 

30 0 

10 

5 

0 

10 

15 

20 

25 

30 

35 

Number of Speakers 

Figure 29 – Space Array DSP usage 

40 

45 

50 

2 

1 

4 

3 

6 

5 

8 

7 

10 

9 

DSP's Used 

• The Space Array provides real time panning of up to 32 

sound sources across multi-speaker arrays containing up to 

50 speakers. 

• Pre-calculated speaker activations allow accurate 

representation of sound in three-dimensions to listeners with 

no sweet spot. 


Features 

Operation 

AniScape 

The AniScape suite is comprised of several simulation tools. 

The three main tools are: 

• The Sound Field Filter, which takes a number of input 

signals and encodes them into a B-format data stream. 

• The Speaker Decoder, which takes B-format encoded data 

and decodes it through a speaker array. 

• The Binaural Decoder, which works in the same manner as 

the Speaker decoder, however B-format data is decoded so 

that it may be used via headphones. 

• Decoding of convolved B-format data to produce real-time 

three-dimensional audio simulations. 

• Easy integration with visualisation systems, utilising SNAP 

messages for network control of source and listener locations 

and positions. 

• Allows the encoding of any number of input signals into Bformat 

audio, dependent on the number of DSP cards 

installed in the system. 

• Virtual spaces or rooms for use in AniScape can be 

generated using acoustic modelling software. 

• Creates Doppler effects for moving sound objects. 

• Provides up to eight sound reflections for each sound object. 

• Allows the decoding of B-format data to headphones or to 

arrays of up to forty speakers. 

AniScape simulations provide a high degree of realism through 

the use of proven acoustic modelling methods coupled with 

Lake Technology’s long, low-latency convolution technology 

(Lake Convolution). AniScape gives total control of the 

locations of multiple sound objects and the listener object 

within a virtual acoustic space. 

Sound objects are encoded into a general three-dimensional 

audio format called B-format by combining source object 

location and angle information, the location and angle 

information of the listener object and the acoustic properties of 


The simulated acoustic spaces used 

in AniScape are implemented in 

real-time using a highly detailed 

room response supplied by a 

modelling packages such as CATT- 

Acoustic, CATT-VRoom (a 

simplified room generation 

package) or Lake Technology’s 

MakeRoom utility. 


the simulated space they inhabit. This generalised format is 

composed of four standard audio signals and may be decoded to 

speakers using the Speaker Decoder or decoded to headphones 

using the Binaural Decoder. The B-format signals may also be 

recorded to multi-track tape or disk and are identical to the 

format produced by a sound field microphone. 


Features 

Operation 

Starting 

The Sound Field Filter 

The Sound Field Filter simulation application is an integral part 

of the AniScape simulation suite. The Sound Field Filter takes 

data from inputs and encodes it into a stream of spatialised Bformat 

data. This data is then filtered through a room response 

to give an output which may be further processed by the 

AniScape Decoder tools, Speaker Decoder or the Binaural 

Decoder, to produce three-dimensional sound simulations. 

• The ability to add a listener and multiple sound objects to a 

three-dimensional Soundscape. 

• The ability to load various acoustically modelled rooms for 

spatial convolution. 

• The ability to set a minimum source distance for unity gain. 

Figure 30 — The Sound Field Filter interface 

The Sound Field Filter must be loaded into the VRack from the 

New/Simulation Tools submenu (for loading instructions see 

the System Tools section of this manual). Once loaded the 


↓ To access the Sound Field Filter 

Editing the Listener Object (Receiver) 

↓ To edit the listener (receiver) 

object 


Sound Field Filter panel will appear in the VRack and a Sound 

Field Filter patchset will appear in the Patch Bay. 

• Click the Sound Field Filter Panel in the VRack. 

The Sound Field Filter interface will appear (Figure 30). 

The Sound Field Filter only provides outputs (In B-format) for a 

single listener object (receiver). 

1. Select the listener object in the Sound Field Filter list. 

2. Click Edit. 


Figure 31 — Sound Field Filter edit/add object dialogue 

3. Type a name for the listener object in the Name box. 



the object will receive SNAP or Locatormessages indicating 

its current (X,Y,Z) position. It is not necessary to set a 

secondary Location Channel number for a listener object. 




transmitted. In the case of the listener object, it represents 

the channel over which headtracking or facing information 

will be transmitted. 

6. Dir Fact (Directivity factor) for the listener object does not 

need to be set, As the listener object does not be generate 

any sound. 

7. Click OK to accept the changes to the listener object 

— or — 

Click Cancel to cancel the changes to the listener object. 


Note: The first two elements in the 

Sound Field Filter object list are 

assumed to be the listener object 

(receiver) and a sound object. As it is 

necessary to have at least these two 

elements to run a simulation, the first 

two objects cannot be deleted. 

Adding and Removing a Sound object 

↓ To add a Sound object to the 

Sound Field Filter list 

For example: if a sound object 

(channel 2) needs to be moved 


sound object (channel 1) its location 






facing is specified as angle channel 

number 1) has a trunk (whose facing 

is specified as channel number 2), the 




elephant, thus the trunk’s angle 

channel number should be defined as 

1 + 2. 


1. Click Add. A dialogue box will appear (see Figure 31 — 

Sound Field Filter edit/add object dialogue). 

2. Type a name for the channel in the Name box. 



the object will receive SNAP or Locatormessages indicating 




4. Set the Secondary Location Channel Number if the object 

is to be moved relative to another object. 




transmitted. 

6. Set the Secondary Angle Channel Number if the object is 


7. Set the Dir Fact (Directivity Factor) of the Sound object. 






d = 0.000 Omni 





↓ To remove the Sound object 

from the Sound Field Filter list 

Editing a Sound Object 

↓ To edit the details of Sound 

objects 

↓ To reset the position of a 

Listener (Receiver) or Sound 

object 


8. Click OK to add the Sound object to the Sound Field Filter 

list 

— or — 

Click Cancel to cancel the addition of the Sound object. 

From the main Sound Field Filter window 

1. Select the Sound object to remove by clicking on the Sound 

object in the Sound Field Filter list. 


The Sound object will be removed from the Sound Field Filter 

list. 

1. Select the Sound object to edit by clicking on the Sound 

object in the Sound Field Filter list. 

2. Edit Name, Location channels, Angle channels and 

Directivity Factors as required. 


—or — 


1. Select the Sound or listener object to edit by clicking on the 

Sound object in the Sound Field Filter list. 

2. Click Reset. 

The sound or listener object location will be set to the origin 

point at 0.00, 0.00, 0.00. 


— or — 



Adding Room Reverberation 

↓ To add room reverberation to 

the simulation 

↓ To insert a room between 

existing rooms 

↓ To Delete a room response from 

the Rooms List 


Room reverberation may be produced using convolution filters 

derived from a room simulation package such as CATT- 

Acoustics, or the Lake Technology MakeRoom program. 

1. Click Rooms. The rooms dialogue box will appear (Figure 

34). 

Figure 32 — The Sound Field Filter Rooms dialogue box 

2. Click Add. 

3. Select a Room Response file (*.GEO) from the file 

selection list. A selection of pre-created *.GEO files is 

present on the software CD supplied with your system under 

the rooms/aniscape directory. 

4. Click OK to Load the Response file. 

— or — 

Click Cancel to cancel the loading of the Response. 

5. Click Close to Close the room dialogue box. 

From the Sound Field Filter Main window. 

1. Click Rooms. 

2. Select an entry from the Room Response list under which 

the room filter is to inserted. 

3. From the Rooms dialogue box Click Insert. 

4. Select a Room Response file (*.GEO) from the file 

selection list. 

5. Click OK to Insert the Room response. 

6. Click Cancel to cancel the insertion of the Room response. 

1. Click Rooms. 

2. Select an entry from the Rooms Response list to delete. 

3. From the Rooms dialogue box Click Delete. 

4. Click OK to delete the Room filter. 

— or — 


Setting Minimum Source Distance 

↓ To set the Minimum Source 

Distance 

DSP Usage 

Summary 


Click Cancel to cancel the deletion of the room response. 

The minimum source distance is the distance at which sounds 

have unity gain (that is, they are not made louder or softer when 

they are within this range). This facility is provided to prevent 

excessive gain — and possible hearing damage to a listener — 

when a sound object comes very close to the listener object. 

• Type the distance in virtual meters into the Minimum 

source distance box. 

All instances of the Sound Field Filter require 2 DSPs for 

reverberation filters. 

Each sound source used in the simulation requires 1 DSP. 

For example, When the Sound Field Filter boots up, it uses 3 

DSPs: two for the reverberation filters and one for the preloaded 

sound source. 

• The Sound Field Filter receives audio from inputs 

characterised as Sound objects, encodes them in B-format, 

and convolves them through a room response to produce Bformat 

encoded output. 

• The Sound Field Filter defines which SNAP or 

Locatorchannels the positional and rotational information of 

each object will be passed over. Positional and rotational 

information can be unique to each object or use a relational 

system. Under the relational system, positional and 

rotational information is passed over a set of linked 

channels. 

• The Sound Field Filter allows the definition of a Listener 

object (receiver) and Sound objects to be used in the 

AniScape suite of simulation tools. 


Features 

Operation 

Starting 

↓ To access the Speaker Decoder 

The Speaker Decoder 

The Speaker Decoder is used as a part of the AniScape suite of 

simulation tools. It decodes B-format signals and provides 

outputs for use in arrays of between two (2) and forty (40) loud 

speakers. 

• The ability to configure an array of between two and forty 

speakers for the playback of B-format sound. 

• The ability to specify the location of the listener within the 

speaker array. 

Figure 33 — The Speaker Decoder interface 

The Speaker Decoder must be loaded into the VRack through 

the New/Simulation Tools submenu (for loading instructions, 

see the System Tools section of this manual). Once loaded the 

Speaker decoder will appear in the VRack, and its associated 

patchset will appear in the Patch Bay. 

• Click on the Speaker Decoder panel in the VRack. 


Adding, Inserting, Removing and Editing a Speaker 

↓ To add a new speaker location 

to the array 

↓ To insert a new speaker into the 

array 

↓ To remove a speaker from the 

array 

↓ To edit an existing speaker 

location 

Tip: It is possible to make multiple 

changes to elements of the speaker 

array, before clicking configure. 


The main Speaker Decoder window will appear (see Figure 33) 

Each speaker must be given a location in terms of its physical 

location in meters from the X,Y,Z origin point at 0,0,0. 

1. Click Add. This will create a new speaker with at default 

location at the origin (0,0,0) point. 

2. Edit the X location of the speaker relative to the origin — in 

meters — in the left-hand speaker location box. 

3. Edit the Y location of the speaker relative to the origin — in 

meters — in the middle speaker location box. 

4. Edit the Z location of the speaker relative to the origin — in 

meters — in the right hand speaker location box. 

5. Click Configure to accept the changes. 

1. Select a speaker from the speaker list, in front of which the 

new speaker is to be added. 

2. Click Insert to insert a speaker into the array. This will 

insert the new speaker and renumber the existing speakers. 

3. Edit the X location of the speaker relative to the origin — in 

meters — in the left-hand speaker location box. 

4. Edit the Y location of the speaker relative to the origin — in 

meters — in the middle speaker location box. 

5. Edit the Z location of the speaker relative to the origin — in 

meters — in the right hand speaker location box. 

6. Click Configure to accept the changes. 

1. Select a speaker from the speaker list. 

2. Click Delete, this will remove the speaker from the list and 

renumber the remaining speakers. 

3. Click Configure to accept these changes. 

1. Select the speaker to edit from the speaker list. 


3. Enter new distances in meters for the X,Y,Z coordinates 

relative to the point of origin (0,0,0) in the speaker location 

boxes. 

4. Click Configure to accept these changes. 


Adding and Editing a Listener 

↓ To position the listener 

Note: It is only possible to alter the 

position of the listener after a speaker 

has been added to the array. 

Testing Speakers 

↓ To check the speaker listing 

against actual speaker locations 

↓ To stop solo play from a speaker 

Restoring a Speaker Array 

↓ To restore a Speaker Decoder 

setup 


The listener may be positioned at any point in the speaker array. 

All positional points in the Speaker Decoder are measured in 

meters from an origin point with the X,Y,Z coordinates of 0,0,0. 

1. Type the listeners X point relative to the origin in meters in 

the left-hand observer location box. 

2. Type the listeners Y point relative to the origin in meters in 

the middle observer location box. 

3. Type the listeners Z point relative to the origin in meters in 

the right hand observer location box. 

4. Click Configure to accept the changes to the listeners 

position. 

To ensure each speaker location is configured correctly the solo 

button can be used to isolate output from a single speaker. 

1. Connect a signal through the Sound Field Filter into the 

Speaker Decoder using the Patch Bay. 

2. Play the Signal. 

3. Open the Speaker Decoder. 

4. Select a speaker from the speaker array. 

5. Click Solo. 

6. The signal will only be sent through the specified speaker. 

• Click Solo, and the Speaker Decoder will function normally. 

When a Workspace (*.WSP) containing a speaker array is saved 

(see the System Tools section of this manual), the Speaker 

Decoder setup will also be saved. 

1. Click Import. 

2. From the File menu select the appropriate *.WSP file from 

which you wish to import the Speaker Decoder setup. 

3. Click OK to import the Speaker Decoder setup. 

— or — 

Click Cancel to ignore the import. 


DSP Usage 

Summary 


The Speaker Decoder uses 1 DSP for every 16 speakers 

rendered. 

• The Speaker Decoder, decodes B-format signals and 

presents outputs to a speaker array of two (2) to forty (40) 

speakers in size. 

• The Speaker Decoder stores the locations of speakers, and 

the location of the real world listener. 

• Locations of speakers in an array can be loaded using the 

import *.WSP function. 


Features 

Operation 

Starting 

The Binaural Decoder 

The Binaural Decoder is used as a part of the AniScape suite of 

simulation tools. It decodes B-format signals and provides 

output for use with headphones. The Binaural Decoder also 

provides facilities for using three-dimensional position tracked 

headphones. 

The Binaural Decoder provides: 

• B-format decoding for headphones. 

• The ability to set gain trim levels for headphones. 

• The ability to select different Head Related Transfer 

Functions (HRTFs) and types of headphones for sound 

playback. 

• Facilities which allow the setting of an angle channel for the 

position tracking of headphones. 

Figure 34 — The Binaural Decoder main interface 

The Binaural Decoder must be loaded into the VRack through 

the New/Simulation Tools submenu (For loading instructions, 

see the System Tools section of this manual). Once loaded, the 

Binaural Decoder will appear in the VRack, and its associated 



↓ To access the Binaural Decoder 

Altering Gain 

↓ To alter the signal gain for use 

through headphones 

Specifying, Changing and Deleting the Channel Angle 

↓ To specify an angle channel for 

the transferral of rotational 

information 

Note: The Binaural Decoder can be 

used in conjunction with the Sound 

Field Filter. It is important that the 

angle channel used for the orientation 

of the listener object (the angle 

channel setting in the Sound Field 

Filter) be different from that set in the 

Binaural Decoder. If the two channels 

are the same, confusing effects will 

result. 

↓ To change the angle channel for 

the transferral of rotational 

information 

↓ To delete the channel angle 


• Click on the Binaural Decoder panel in the VRack. 

The main Binaural Decoder window will appear 

(see Figure 34). 

Alter the value of the gain by either: 

1. Typing a value into the Gain box. 

— or — 

Click the Arrows to select the appropriate gain value. 

2. Click Configure to accept the changes made to the gain 

levels. 

The angle channel selects the channel along which rotational 

information from a tracking device will be passed. 

1. Type a number, other than zero, into the angle channel box. 

2. Click Configure, this will set the angle channel to the 

number provided. 

1. Type the new channel number into the Angle Channel box. 

2. Click Configure, this will set the angle channel to the 

number provided. 

1. Set the Angle Channel box number to 0. 

2. Click Configure, this will halt the transmission of 

information through the Angle Channel. 


Selecting and Changing a Head Related Transfer Function (HRTF) 


Head Related Transfer Functions (HRTFs) are sound filters 

which mimic the signal path from a sound source to the 

eardrums of the listener. 

The HRTFs used in the Binaural Decoder are read from a file 

stored in the directory Directory\Binaural\hrtf where 

Directory is defined by the path to the Huron 3.2 software 

directory. 

HRTF Description 

armhoriz.sim B-format Decode, Horizontal only, based on HRTF 1 

armperi.sim B-format Decode, Including height, based on HRTF 1 

sgbhoriz.sim B-format Decode, Horizontal only, based on HRTF 2 

sgbperi.sim B-format Decode, Including height, based on HRTF 2 

↓ To alter the HRTF used by the 

Binaural decoder 

Selecting Headphones 

↓ To alter the Headphone 

Response 

DSP Usage 

1. Select a New HRTF from the HRTF list. 

2. Click Configure to confirm the selection. 

Each type of headphone has a different response to various 

frequencies. 

The headphone equalisation filters used in the Binaural Decoder 

are read from files stored in the directory 

Directory\Binaural\Headphns where Directory 

is defined by the path to the Huron directory (usually 

C:\Huron32\). 

Currently the Binaural Decoder is supplied with a generic type 

of headphone. 

1. Select a New Headphone Response from the Headphone 

List. 

2. Click Configure to confirm the selection. 

The Binaural Decoder uses 1 DSP. 


Summary 


• The Binaural Decoder decodes B-format signals and 

processes them for listening over headphones. 

• The Binaural Decoder allows the setting of HRTF and 

headphone responses to provide clearer and more effective 

three-dimensional sound sourcing. 


Features 

Operation 

MultiScape 

The MultiScape VRack tool is a self-contained threedimensional 

sound modelling application, which allows multiuser 

real time acoustic simulations. 

• Provides facilities for the control of sound objects via the 

SNAP interface. 

• Allows the creation of multi-user soundscapes, including 

door effects, room reflections, and live feeds 

• Produces room reverberation by using room filters derived 

from room acoustic modelling programs such as CATT- 

Acoustic, CATT-VRoom or Lake Technology’s MakeRoom 

software. 

• Provides the output of the simulation to each user over 

normal or tracked headphones or arrays of up to 12 speakers. 

MultiScape allows users to participate in highly realistic realtime 

three-dimensional acoustic simulations with multiple 

moving sound objects, multiple simulated rooms and multiple 

listener objects. Rooms are simulated using Lake’s proprietary 

convolution technology — Lake Convolution — which gives 

realistic room reverberation and accurate localisation. 

Playback can be through either tracked headphones or an array 

of between two and twelve speakers. 

MultiScape integrates with other applications via the VRack. 

Using MultiScape through the VRack interface allows a variety 

of processing options such as equalisation (using Lake EQ), to 

be inserted into the processing of MultiScape sound objects. 

All MultiScape functions are controllable via SNAP messages. 


Starting 

↓ To access MultiScape 

Figure 35 — The MultiScape main interface 

Adding Editing and Removing Sound Objects 

Adding a Sound Object 

↓ To add a sound object 


MultiScape must be loaded into the VRack from the 

New/Simulation Tools submenu. (for loading instructions, see 

the the System Tools section of this manual). Once loaded the 

MultiScape panel will appear in the VRack and a MultiScape 


• Click the MultiScape Panel in the VRack. 

The MultiScape Interface will appear (Figure 35). 

The MultiScape main interface object list, contains the list of 

sound objects, speaker arrays and headphones associated with 

the current MultiScape simulation. 

A sound object in MultiScape characterises a source of input to 

the Huron system, and can include a wide range of sound 

generation devices. The sound object can be given; a Name, 

Location Channel, Angle Channel, Directivity Factor, and a 

maximum Sound Pressure Level (SPL). Once characterised, a 

sound object will appear as an input patch point on the 

MultiScape patch panel in the Patch Bay application, where it 

can be patched to an audio source. 

• Click Add. 

The following dialogue box will appear (Figure 36). 


Figure 36 — The MultiScape add sound object dialogue 


d = 0.000 Omni directional 






1. Select Sound. 

2. Type a Name for the object in the Name Edit Box. This 

name will appear in the Patch Bay next to the appropriate 

patchpoint after the sound object has successfully been 

added. 



the object will send and receive SNAP or Locatormessages 

indicating its current (X,Y,Z) position. 




transmitted. 

5. Set the Dir Fact (Directivity Factor) of the Sound object. 




6. Set the Peak Sound Pressure Level (SPL). The Peak SPL 

is measured in Decibels and restricts the maximum sound 

pressure that a Sound object can produce. 

7. Click OK to accept the configuration of the Sound object. 

The Sound object will be added to the MultiScape object list, 

and the appropriate patchpoints will be added to the 

patchpanel in the Patch Bay. 

— or — 

Click Cancel to Cancel the addition of the Sound Object. 


Editing a Sound Object 

↓ To edit the configuration of a 

Sound object 

Removing a Sound Object 

↓ To remove a Sound object 

Adding Editing and Removing Speaker Arrays 

Adding a Speaker Array 

↓ To add a speaker array 


1. Select a Sound object from the MultiScape List. The Sound 

object will be highlighted. 


The Sound object Dialogue box will appear (Figure 36). 

3. Edit the required fields. 

4. Click OK to save the configuration changes. 

— or — 

Click Cancel to Cancel the changes made to the Sound 

object. 

1. Select the Sound object to remove from the MultiScape 

List. 


The Sound object will be deleted from the MultiScape List. 

Speaker Array objects in MultiScape represent real-world 

speaker arrays designed for the playback of audio simulations to 

a listener. 

Speaker arrays in MultiScape are handled via the MultiScape 

list speaker dialogue (Figure 37). Each Speaker object 

represents an array of up to 12 speakers, about which virtual 

sound objects can be panned. 

1. Click Add. 

2. Select Speaker. 

The MultiScape object dialogue box will expand to include a 

series of speaker related options (See Figure 37). 


Figure 37 — MultiScape add speaker object dialogue 

Speaker Pan Mode 


3. Type the number of speakers in the array into the Speaker 

number box. The speaker box will expand to accommodate 

up to 12 speakers. 

Each speaker must be given a location in terms of its physical 

location in meters from the X,Y,Z origin point (and the location 

of the listener) at 0,0,0. 

4. For each speaker in the array; 

• Edit the X location of the speaker in the left-hand speaker 

location box. 

• Edit the Y location of the speaker in the middle speaker 

location box. 

• Edit the Z location of the speaker in the right hand speaker 

location box. 

5. Check Input if the Speaker object will have an input to the 

system (for example, the user of the array is using a 

microphone). If input is checked, an appropriate patch point 

will appear in the MultiScape patch panel in the Patch Bay. 

6. Type the Sensitivity of the listener. This value given in dB 

specifies how sensitive the real world listener is to sound. A 

high number indicates a low sensitivity and the gain of 

sound sources will be scaled upwards accordingly. A value 

of 100 is normal sensitivity. 

7. Select a Mode for the speaker array to operate in. Speaker 

arrays in MultiScape can be configured to operate in three 

modes: 

• The first is as a Speaker Panner, which will pan virtual 

sound objects about the speaker array. 

• The second is as a B-format decoder that decodes B-format 

sound and plays it to the appropriate speakers in the array. 

• The third mode encodes sound object movements and 

outputs them as B-format sound. 

When Speaker Pan mode is selected, further panning options 

will appear. These options are: 


B-format Decode Mode 

B-format Mode 

Editing a Speaker Array Object 

↓ To edit the configuration of a 

Speaker Array object 

Removing a Speaker Array Object 

↓ To remove a Speaker Array 

object 


• Constant power panning which will maintain the same total 

output of power from the speaker array regardless of the 

listeners distance from the speaker. 

• Constant amplitude panning which will provide a constant 

volume to a listener, by altering the power output to the 

speaker. 

• Warp Factor which effects how sounds pan between 

speakers. A Warp factor of 1.00 gives a linear pan between 

two speakers. For a value greater than 1.00 the sound will 

tend to stick to speakers panning rapidly between them. 

With values less than 1.00, The sound will tend to linger 

between the speakers before moving on. 

B-format decoding decodes B-format sounds to an array of 

speakers. 

This mode produces output which is B-format encoded. The 

X,Y,Z,ω signals can be utilised by other applications or 

products which undertake their own B-format decoding. 

• Click OK to confirm the addition of the speaker array object 

to the MultiScape object list. 

— or — 

Click Cancel to cancel the addition of the speaker array 

object. 

1. Select the Speaker Array object from the MultiScape List. 


3. The Speaker Array dialogue box (Figure 37) will appear. 

4. Edit the required fields. 

5. Click OK to save the configuration changes. 

— or — 

Click Cancel to cancel the changes made to the Speaker 

Array. 

1. Select a Speaker Array object from the MultiScape List. 


The Speaker Array object will be deleted from the MultiScape 

List. 


Adding Editing and Removing Headphone Objects 

↓ To add a headphone object 


Headphone objects in the MultiScape list represent a pair of 

headphones used by a listener to which the binaural decoding of 

the sound simulation will occur. 

1. Click Add. 


box. This location channel will be the channel along which 

messages are passed which pinpoint the position of the 

virtual listener object in the Soundscape. 

3. Select the headphone radio button. 

A set of new options related to headphone decoding will be 

shown. 

Figure 38 — MultiScape add speaker array dialogue 

4. Type a Name for the object, eg. Listener_1. 

5. Select a Head Related Transfer Function from the HRTF 

list. This list is comprised of the following HRTFs: 

HRTF Description 

armhoriz.sim B-format Decode, Horizontal only. 

armperi.sim B-format Decode, Including height. 

armpan.sim Panning mode, MultiScape. 

sgbhoriz.sim B-format Decode, Horizontal only. 

sgbperi.sim B-format Decode, Including height. 

sgbpan.sim Panning mode, MultiScape. 

transaur.sim Transaural HRTF. 

6. Add an Angle channel over which facing and angular 

information will be passed (this option is used primarily for 

headtracked headphones). 

7. Check the input box if the headphone user has an input. For 

example a microphone, once checked an input patch point 



Editing Headphone Objects 

↓ To edit a headphone object 

select the object in the 

MultiScape list 

Removing Headphone Objects 

↓ To remove a headphone object 

from the MultiScape object list 

MultiScape Rooms 


8. Type the sensitivity of the listener. This value given in dB 

specifies how sensitive a user is to sound. A high number 

indicates a low sensitivity and sound sources will be scaled 

upwards accordingly. A sensitivity of 100 is considered 

normal. 

9. Click OK to enter the Headphones and their configuration 

into the MultiScape Object List. 

— or — 

Click Cancel to Cancel the addition of the Headphone object 

to the list. 


2. Edit the Headphone objects settings. 

3. Click OK to accept changes. 

— or — 


1. Select the Headphone object in the MultiScape Object List. 


The headphone object will be deleted from the list. 

Figure 39 – MultiScape room definition 

MultiScape provides the capability of having multiple 

interconnected rooms with different room impulse responses. It 

also allows for doors and door effects to be set up between 

rooms. All these effects may be altered in real time (ie. closing 

doors etc.) 

Virtual rooms are defined using two characteristics. The first 

characteristic is the room property. Properties are defined by 


Room Map Function 

↓ To access the room map 

function 

↓ To view the attributes of a room 

↓ To close the Room map dialogue 


the impulse response of the room. These impulse responses 

produce the reverberant effects in the virtual room. The second 

property is the size of the virtual room. In MultiScape the size 

of the virtual room is defined by a room segment. Each room 

segment is a rectangular based prism (cuboidal or hexahedron), 

with a lower left X1,Y1,Z1 coordinate and an upper right 

X2,Y2,Z2 coordinate. 

A MultiScape room simulation may have multiple room 

segments all of which may have the same or different room 

properties, and multiple doors. 

Door Simulations in MultiScape are based on rectangles, on the 

boundary between room segments. 

The room map function shows the dimensions, and attributes of 

doors and rooms utilised in the current MultiScape simulation. 

Each room and its associated doors, and room response 

attributes are displayed sequentially in the room map. 

• Click Map the following window will appear (Figure 40). 

Figure 40 — MultiScape’s room map dialogue 

• Click the Room Spin box until the desired room appears in 

the Map window. 

As Rooms, Doors and Room properties are added to the 

simulation they will appear in the appropriate section of the 

Room Map. 

• Click Close. 


Adding Rooms, Deleting Rooms 

Adding Room Properties 

Creating Room Segments 

Associating Room Properties with Room Segments 

Creating Assigning and Using Doors 

MultiScape Simulation Options 


Rooms must be loaded into MultiScape via SNAP or Locator 

scripts. 

The procedure outlined below illustrates the five-step process 

involved in creating a virtual Soundscape in MultiScape. 

Room properties define the impulse response of a room. These 

responses can be measured (using Lake Technology’s 

measurement application), or generated from simulated 

soundspaces using modelling software such as the CATTacoustics 

room modelling package or Lake Technology’s 

MakeRoom application. 

Create a room segment (a virtual space). Three-dimensional 

spaces may be divided up into a series of rectangular based 

prisms. To define complex room geometries or spaces a series 

of these prisms may be linked together using the same room 

properties (impulse responses). 

Once defined, the room properties or impulses may be 

associated with the room segments, to give the virtual acoustic 

spaces realistic sound properties. Each segment can have a 

single property, but multiple room segments may be associated 

with the same property. 

Doors are created as rectangular objects with lower left, and 

upper right X,Y,Z coordinate pairs and have to be attached to 

specific room segments. Doors must be placed at the junction 

of two rooms. Once doors are associated with room segments it 

is possible to give a value representing the relative value of 

their closure. Doors with a value of 0.00 are fully closed while 

doors with a value of 1.00 are fully open. 

MultiScape provides three levels of simulation. 


No Room simulation 

↓ To select No Room simulation 

Room Simulation 

↓ To select Room Simulation 

Door Simulation 

↓ To select Door Simulation 

DSP Usage 


When the room simulation option is not checked MultiScape 

acts as if sound objects and listener objects were placed in a 

non-reverberant room of unlimited size. The spatialising of 

sounds, and the apparent movement of Sound objects will still 

occur. 

• Uncheck the Room Simulation option. 

When Room Simulation is checked, any defined room segments 

and their associated properties will come into effect. Any doors 

which have been defined will operate in a simplistic manner, 

behaving as if they were a point source with the volume 

dependent on the distance of sound objects from the door and 

the relative closure of the door. 

• Check the Room Simulation option. 

Door simulation can only be enabled when Room Simulation is 

turned on. With door simulation checked, extra DSP resources 

are utilised to make door effects more realistic. When this 

option is turned on, doors allow sounds to be processed using 

room impulses. Up to sixteen doors can be added for each DSP 

used. 

• Check the Door Simulation option (this option is only 

available when Room Simulation is checked). 

Multiscape’s DSP requirements are illustrated in the following 

table: 

Each user 

requires 

Each room 

requires 

Door mixer 

(all doors) 

No room / 

door 

simulation 

Room 

simulation 

only 

Room and 

door 

simulation 

1 DSP 3 DSP 1 DSP 

— — 2 DSP 

— — 1 DSP 


Summary 


• MultiScape provides an integrated set of tools for multiple 

people multi-room, multi-user acoustic simulations. 

• It provides the ability to add Sound objects, Speaker Arrays 

and Headphones to a virtual Soundscape. 

• It provides fully featured room and door effects, which can 

be updated in real-time. 


Features 

Operation 

HeadScape 

HeadScape is a subset of the simulation tools used for accurate 

headtracked auralisation. HeadScape allows the 

implementation of highly accurate localisation of sounds over 

headphones. 

• HeadScape utilises a filter switching technique to provide 

accurate aural simulations controlled by a headtracked pair 

of headphones. 

• HeadScape uses responses derived from acoustic modelling 

software in order to obtain accurate room responses for use 

in simulations. 

HeadScape is Lake Technology’s premier head-tracked 

auralisation system. Designed for critical evaluation of room 

acoustic simulations, HeadScape makes use of Lake 

Technology’s proprietary low-latency long convolution 

technology to switch long FIR filters smoothly and efficiently 

in response to the listener’s head movement. 

To create a consistent and realistic simulation HeadScape 

employs a two stage filtering process. The late part of the 

room’s impulse response is kept fixed but the early part is 

selectively switched depending on the listener’s head position. 

The early part is of the order of 4000 samples long and includes 

a binaural representation of the early reflections in the space. 

This feature enables HeadScape to create virtual sound objects 

that are strongly and clearly externalised. 


Starting 

↓ To access HeadScape 

Setting the Angle Channel 

↓ To set the angle channel number 

Figure 41 — The HeadScape main interface 


HeadScape must be loaded into the VRack through the 

New/Simulation Tools submenu. (for loading instructions see 

the System Tools section of this manual). Once loaded, The 

HeadScape rack panel will appear in the VRack, and its 

associated patchset will appear in the Patch Bay. 

• Click on the HeadScape panel in the VRack. 

The HeadScape main interface will appear (Figure 41). 

The Angle channel is the channel over which the orientation of 

the listener is transmitted. In a HeadScape simulation this 

channel is most often utilised by a headtracking device such as 

the Polhemus InsideTrak system communicating using SNAP 

commands. 

• Type the Angle Channel Number in the Angle edit box. 

Adding the Environment Related Transfer Functions (ERTF’s) 

↓ To load the ERTF’s into 

HeadScape 

The ERTF’s in a HeadScape simulation represent the room 

reverberation or the tail part of the composite HeadScape filter. 

By changing these filters it is possible to alter the type of room 

the sound appears to be generated in. 

1. Click Browse … next to the Left edit box. The following 

dialogue box will appear. 


Figure 42 — HeadScape’s add ERTF’s dialogue box 

Adding, Editing and Deleting Early Impulses 

↓ To add a series of front impulses 

to the list 

↓ To edit (swap) a series of front 

impulse filters in the list 

↓ To delete a series of front 

impulses from the list 


2. From the Select *.SIM file window, Choose the left-hand 

impulse response file for the room required. 

3. Click OK to load the left Impulse response filter. 

4. Click Browse … next to the Right edit box, and repeat the 

procedure above to load the right side impulse. 

The switching of the early part of the impulse response in a 

HeadScape simulation provides the means of clearly and 

accurately externalising sound sources. A series of filters is 

loaded into the HeadScape tool via a descriptive HeadTurn file 

(*.htr) — for more details on the HeadTurn file see the file 

description below. 

1. Click Add. 

2. Select a the relevant *.htr file. 

3. Click OK. 

1. Select the *.htr file you wish to swap for another. 


3. Click OK, the selected *.htr file will be replaced. 

1. Select the *.htr file you wish to remove. 


Configuring HeadScape 

↓ To configure HeadScape 

Note: Configuration is required each 

time any of the HeadScape 

Parameters change, this is illustrated 

by the un-dimming of the Configure 

button when changes have been made 

to the HeadScape setup. 

Saving 

HeadScape files and the HeadScape File Format 


2. Click Delete, the selected *.htr file will be removed from the 

list. 

Once the angle channel, front impulses and end impulses have 

been selected the HeadScape application needs to be 

configured. Configuration entails the booting of DSPs and 

loading of filter series into the DSP card’s memory to allow 

realtime switching of early impulses. 

• Click Configure. 

HeadScape will load the required DSP filters and load the 

impulse responses into memory. Once HeadScape is 

configured the Configure button will dim. 

Once configured and saved in a workspace, the current 

configuration of HeadScape will be restored when that 

workspace is loaded. 

HeadScape uses a collection of impulse responses in *.SIM files 

to represent an acoustic space. Acoustic modelling programs 

(such as CATT Acoustic) have the ability to generate these SIM 

files automatically. A pair of short *.SIM files exist for each 

angle of head rotation for a particular listening position. 

Another pair of long SIM files exist which represent the 

reverberant tail of the impulse response for the room. 

The relationship between the short SIM files and the angles of 

head rotation they represent is defined in a “HeadTurn” (*.htr) 

file. The HeadTurn is a text file containing a list of numbers 

with the following format 

aaa.a eee.e 

aaa.a represents an angle from 0 to 360 degrees with 0.5 degree 

resolution. This angle is the angular position of the head for the 

specified source-receiver simulation. 

eee.e represents an elevation from +180 to -180 degrees with 

0.5 degree resolution. This angle is the elevation of the head for 

the specified source-receiver simulation. 


HURON1 

3840 

0.0 0 

1.5 0 

3.0 0 

4.5 0 

6.0 0 

7.5 0 

9.0 0 

10.5 0 

12.0 0 

Angle Elevation SIM files represented 

0.0 0 h000180l.sim and h000180r.sim 









Angle Elevation SIM files represented 

0.0 -11.5 h000157l.sim and h000157r.sim 


0.0 11.5 h000203l.sim and h000203r.sim 







DSP Usage 


Each number represents a pair of SIM files for the specified 

angle. The SIM file names will be AAAEEEl.sim and 

AAAEEEr.sim, where AAA=2 x aaa.a and EEE=180 + (2 x 

eee.e.) 

For example, below is the first 10 lines of an example 

HeadTurn file. 

HURON1 specifies the version number of the HeadTurn file. 

3840 represents the length of the head SIM files created. If this 

length is greater than the maximum head filter length, then 

HeadScape will truncate this length to its maximum (of 3585). 

Each of the following numbers represent the SIM file pair for 

the specified angle. The relationship between the numbers in 

the HeadTurn file and the actual SIM files is as follows: 

A HeadTurn file can also specify elevations, as follows: 

HeadScape uses 2 DSPs for the tail filters representing the room 

response, and 2 DSPs for each sound source in the simulation. 


Summary 


• HeadScape is a tool designed for the accurate and clear 

externalisation of sound sources for research purposes. 

• HeadScape utilises a two stage impulse response system to 

ensure accuracy. 

• HeadScape uses a simple paired text file (*.htr) to map the 

early part of the impulse response to representative 

directions with a resolution of up to 0.5 degrees. 


Features 

Operation 

Sonic Animator 

Sonic Animator gives sequencing capability to Huron messages 

in the way that audio or MIDI sequencing applications give 

sequencing capability to audio recordings or MIDI messages. 

In particular, this initial Sonic Animator version allows sound 

source location sequences to be recorded, replayed, and 

generally manipulated in a graphical environment. 

Sonic Animator is not a VRack application, but stands alone 

and communicates with VRack applications such as the Space 

Array through Huron messages. 

• Recording and playback of sound source location sequences 

using either the internal PC clock or external MIDI Time 

Code (MTC) through an ordinary sound card in the Huron. 

• A track-based temporal view where each track corresponds 

to a Location Channel Number. 

• Temporal manipulation of location sequences, such as cut, 

copy, paste, clear, merge, split, shift by. 

• Importing and exporting of Locator scripts, allowing old 

performances to be used without having to be recreated. 

Locator scripts are described in the Locator subsection of 

this manual. 

• Playback with linear interpolation between location 

messages. Thus, rather than specify every location message 

to be sent, the user may specify extremes and let Sonic 

Animator interpolate between them. 

• An internal resolution of 1 millisecond and an internal 

playback or record resolution of 10 milliseconds. 

In order to successfully use Sonic Animator, an understanding 

of the hierarchical data model is required. The model is 

represented pictorially in the following diagram. 


Scene 

(List of Tracks) 

Track 

(List of Trajectories) 

Trajectory 

(List of Events) 

Event 

Figure 43 – Sonic Animator Hierarchical Model 


The highest level element is the scene. A scene corresponds 

logically to a single soundscape that has a start time and a 

duration. It contains a number of tracks that will be played 

back simultaneously. 

A track is a conceptual device that describes all movement of a 

single sound source. Each sound source will be replayed on a 

Location Channel Number so each track has its own unique 

channel mapping. 

A trajectory is a conceptual device that represents a single 

movement sequence within a track. A trajectory has a defined 

start time and duration. A track may consist of multiple 

trajectories if the trajectories are logically separate movements. 

For example, the sound of a car may approach from a distance 

and then stop in front of the listener. Some time later, it may 

turn around and drive off again. The two movements can be 

recorded separately and dealt with as separate trajectories. 

An event is a location message that occurs at a defined time 

within a trajectory. Events are stored relative to the start time 

of the trajectory. This means that the first event is always at 

time = 0 and the last at time = duration. The duration is defined 

by the last event in the trajectory. 

An event describes the location of a sound source at a defined 

instant in time. Thus it has a duration of 0 and can be defined 

by just a start time. Conversely, scenes, tracks, and trajectories 

have start and end times. 

Trajectories cannot overlap within a track. Similarly, multiple 

events cannot occur at the same time within a trajectory. 


Starting 

↓ To create a new scene 


Figure 44 represents a scene, contains 9 tracks, 6 trajectories, 

and an indeterminate number of events. 

Figure 44 – Sonic Animator Interface 

Sonic Animator must be started from the Windows 2000 Start 

menu. It is a stand alone application and does not require the 

VRack to be running. 

A new scene must be created before trajectories can be recorded 

or Locator scripts imported. 

• Select New from the File menu and the Scene Properties 

dialog will appear allowing modification of the default scene 

settings. 


Figure 45 – Scene Properties dialog 


The Description and Author fields allow descriptive 

information that describes the scene for other users. 

The Timeline format radio buttons affect the format of the 

timeline at the top of the screen. When the SMPTE format is 

selected, the user is allowed to choose from one of the defined 

MTC frame rates. This option affects the display of the 

timeline only and has no affect on sequencing rates. 

The Internal timer will use the PC clock for timing during play 

and record, and the user may select a Sample Rate in 

milliseconds. The External timer will use one of the MIDI 

devices for synchronisation. 

The Insertion mode affects the way trajectories behave when 

they overlap during manipulation operations. Don’t Overwrite 

will clip the overlapping trajectory, leaving the original 

untouched; Overwrite will always overwrite the overlapping 

section of the original trajectory; while Prompt will allow the 

user to choose whether to overwrite whenever an overlap 

situation occurs. 

There are three ways of inserting event information into a Sonic 

Animator Scene. The first is to import a previously created 

Locator script. Another is to record Huron messages directly 

using a tools such as Locator or Tracker. The third is to directly 

edit the ASCII Scene file in a text editor. These three methods 

are described below. 


↓ To import a Locator script 

↓ To record messages from 

another application 

↓ To edit a scene file 


1. Select Import from the File menu. 

2. Select a file to import from the file dialog. 

3. Select the time you wish to paste at. If the Ignore empty 

time option is selected then the first event in the script will 

be inserted at the given time. Otherwise the first event will 

be pasted with whatever offset it has within the script. 

The import mechanism will read the script and attempt to match 

Locator channels with existing tracks in the current scene. If a 

channel is found in the script that doesn’t match the channel of 

any existing track in the scene then a new track will be created 

with the desired channel. 

Within a Locator script, the events mapped to each channel are 

assumed to correspond to a single trajectory. So Locator scripts 

can’t contain multiple trajectories within a track. 

Huron tools such as Locator and Tracker create messages that 

can be recorded by pressing the record button in Sonic 

Animator. This section assumes that internal synchronisation is 

used. See the section on External MTC Synchronisation for 

external timing. 

When the record button is pressed, Sonic Animator listens for 

messages indicating that object locations are changing, and any 

messages whose Location Channel Numbers match the channel 

number of an existing, armed track will be inserted into a new 

trajectory in that track. A message destined for a track that isn’t 

armed or doesn’t exist in the scene will be ignored. 

The Scene file format uses ordinary ASCII characters and can 

be edited using any text editor. The file format uses the same 

hierarchical model as the rest of the application. 

The first three words of the file must be Sonic Animator Scene 

and the next two must be the version information. 

After this, the file is organised into keywords and corresponding 

data. Most keywords have one data element. See the file 

Sample.scene for an example of the allowable keywords and 

refer to the Command Reference section of this manual for a 

complete list. 

The LocationTrack keyword must be followed by a block 

section containing all the information for a track. When the file 

is parsed, the tracks will be inserted into the scene in the order 

they appear in the file. 

Similarly, track sections have a set of allowable keywords, one 

of which is LocationTrajectory. Note that the start time of a 

trajectory is specified relative to the start of the scene. Within 

the LocationTrajectory block are the trajectory information 

and a list of events. The data for the LocationEvent keyword 

consists of a time relative to the start of the trajectory and a 

three dimensional Cartesian position. 

The keywords within a section may appear in any order and any 

unrecognised keywords will be ignored. Comments are 


Manipulating Tracks 

↓ To add a track 

↓ To change a track’s position in 

the scene 

↓ To delete a track 


allowed, with any text on a line following a semicolon (;) being 

ignored. 

• Select Add Track from the Track menu 

— or — 

Double click on an empty part of the scene. 

The Track Properties dialog will appear asking you for 

information about the new track. 

Figure 46 – Track Properties dialog 

The first free channel number will appear as the default for the 

new track. Select Interpolation if you wish this track’s output 

to be a linear interpolation between the nearest two events 

within a trajectory. The Solo and Mute functions allow you to 

silence all other tracks, or just this one during playback. The 

Arm feature specifies whether to add events to this track during 

record. If Arm is unchecked then events on the track’s channel 

will be ignored when recording. 

1. Select Move Track from the Track menu. 

2. Identify the desired track, either by numerical position 

within the scene, or by name. 

3. Select the new desired position and click OK. 

1. Select Delete from the Track menu. 

2. Identify the desired track, either by numerical position 

within the scene, or by name and click OK. 


↓ To change a track’s properties 

Manipulating Trajectories 

↓ Selecting trajectories 

↓ Cutting, copying, and pasting 

trajectories 

↓ To clear a trajectory 

↓ To merge two or more 

trajectories 

↓ To split a trajectory 

• Select Properties from the Track menu. 


• Select trajectories by clicking on them with the left mouse 

button. Use Shift or Ctrl to select and unselect multiple 

trajectories. 

— or — 

Click the left mouse button on an empty piece of track and 

drag the focus rectangle over the trajectories you want selected. 

These operations use the standard Windows clipboard to 

serialise and retrieve trajectories and are available either from 

the Edit menu or by right-clicking on the desired trajectory. 

When a trajectory is cut or copied, a copy of its parent tracks is 

also made. This serves two purposes: the first is to provide a 

record of where the trajectory originated from; the second is to 

have a record of the track properties, in case a new track will be 

created to paste into. 

If there are trajectories selected when the paste command is 

issued then the clipboard trajectories will replace the selected 

ones. Conversely, if no trajectories are selected then the Paste 

dialog will appear, prompting for a track and a time to begin 

pasting from. This will be the new position of the first 

trajectory on the clipboard. The first trajectory is always the 

earliest selected trajectory in the first track containing selected 

trajectories. 

If trajectories will be pasted past the end of the scene then the 

scene will be extended to accommodate them. This does not 

apply past the beginning of a scene. Similarly, if the 

trajectories will be pasted past the limit of tracks then more 

tracks will be created. 

• Select Clear from the Edit menu. 

• Select Merge from the Edit menu. 

Merging two or more trajectories is allowed when the 

trajectories are located on the same track and when there are no 

unselected trajectories between those to be merged. 

1. Select Split from the Edit menu. 

2. Enter a time at which to split the trajectory. The time must 

be within the selected trajectory. 

The trajectory will be split into two, with all events up to and 

including the split time going into the first new trajectory and 

all remaining events going into the second. 


↓ Exporting a trajectory to a 

Locator script 

↓ To change a trajectory’s 

properties 

↓ Manipulating trajectories by 

clicking and dragging 

Sequencing Functions 

↓ To play the current scene 

↓ More on recording 


• Select Export from the Edit menu and then choose a 

filename for the script in the file dialog. 

The trajectory will be written into a new Locator script, 

beginning at time 0. 

• Select Properties from the Edit menu. 

As well as using the formal methods of trajectory manipulation, 

it is also possible to move trajectories by clicking on them and 

dragging them to a new desired position. Use the Ctrl key 

while releasing the mouse to copy the selected trajectories 

rather than move them. 

• Click the Play button or select Play from the Sequence 

menu. 

If using internal synchronisation, the scene will begin playing 

from the time marker position. Sonic Animator uses the PCs 

internal timer mechanisms to generate a timer message 

whenever the interval specified in the Scene Properties dialog 

has elapsed. 

At this stage, two things may happen. If a track is interpolated 

and the current time is corresponds to somewhere inside a 

trajectory, then the sequencer will find the two events on either 

side of this time and linearly interpolate between them to give a 

position to output. If the time does not fall within a trajectory 

then no messages will be sent and the location will remain 

where it was at the end of the last trajectory. Conversely, if the 

track is not interpolated, then the sequencer will play all events 

that have become due since the last timer message. 

The time marker will move along as the scene is played. 

However, updating the time marker is a low priority task and 

only happens when the system is idle. Thus, when the 

sequencer is using a lot of CPU time sending events, the time 

marker won’t move. 

Recording uses a similar mechanism to playing but is slightly 

more complex. The same timer mechanism is used to update 

the current time and the time marker. When a message arrives, 

the sequencer finds the time of the last timer message and 

applies this to the received message when writing it into the 

scene. If multiple messages arrive between timer updates then 

the later events are discarded. 


External MTC Synchronisation 

↓ Connecting a MIDI device 

↓ Sequencing under external 

synchronisation 

User Interface Features 

↓ Zooming in and out 

↓ Using the timeline 

DSP Usage 

Summary 


An external MIDI Time Code generating device such as the 

Opcode Studio 64XTC may be connected to the Huron through 

the MIDI port on the Huron’s sound card. The Opcode device 

is driven by a PC or Macintosh through a serial cable. 

• Select External in the Scene Properties dialog. 

When playing using an external timer, the timer message 

updates are provided by the external device rather than the PC 

clock. The same interpolation and non-interpolation behaviour 

occurs. 

Similarly, when recording under an external timer, the timer 

message is used to set the current time and messages that arrive 

are assigned the most recent time. 

To zoom in and out on a scene, use the buttons on the Zoom 

Toolbar. Alternatively, select Zoom In or Zoom Out from the 

View menu. Zooming in or out will adjust the view by a factor 

of 2 in either direction and zooming in will centre the new view 

on the middle of the old view. 

The timeline resides at the top of the scene window and can be 

moved to the bottom of the window by right-clicking on it and 

selecting either Dock->Top or Dock->Bottom. 

Clicking on the timeline will also move the position of the 

current time marker. 

Sonic Animator does not use DSP resources. 

• Sonic Animator is a real-time sound source location 

sequencer for the Huron. In this manner, it replaces some of 

the functionality of Locator. 

• The familiar sequencer interface allows trajectories of sound 

sources to be manipulated with ease. 

• Sonic Animator will synchronise to MIDI Time Code for 

full integration with existing sound equipment. 


Other Simulation Tools 

This sub-section describes the other simulation tools provided 

by Lake Technology for their Huron systems. 

These tools can be used in association with other VRack 

applications to provide an enormous range of sound processing 

configurations. Some of the more commonly used 

configurations are described below: 

Locator: used to send positional information for sound objects 

in AniScape, MultiScape, and Space Array simulations. In 

addition, it provides a simple scripting interface through which 

rooms and acoustic spaces can be configured in AniScape, 

MultiScape, and Space Array. The Locator can also be used to 

load, associate, and play wavetable sounds through the 

WavePlayer application. Finally, the Locator provides facilities 

for sending MIDI messages and program control messages to 

the Convolver. 

WavePlayer-2: An enhanced version of WavePlayer featuring 

large sample playback and zero DSP load. 

WavePlayer: primarily used in AniScape, MultiScape, and 

Space Array simulations, and provides a facility for playing 

*.WAV files (converted to the *.SIM file format) to be used as 

input to other applications. 

Tracker: primarily used in headtracking applications such as 

HeadScape. It can also be used to position sound objects in 

AniScape, MultiScape, or Space Array simulations, or be used 

to input trajectories into Sonic Animator. 

Socket: when used in conjunction with the SNAP Spatial 

Network Audio Protocol, the Socket interface can be used to 

control almost every aspect of a Huron from a remote 

workstation. 

Makeroom: The make room utility is a stand-alone Windows 

application with the ability to create room impulse response 

filters for use in Convolver, AniScape, MultiScape, and Space 

Array simulations. 


Features 

The Locator 

The Locator tool provides a graphical interface for moving 

multiple AniScape, MultiScape, or Space Array Objects in a 

two-dimensional Plane (X,Y). It also provides facilities for 

controlling Lake Technology applications such as the 

WavePlayer. The Locator tool is designed as a testing and 

configuration verification program only. It is not intended to be 

used in critical applications where precise, synchronised control 

over sound objects, listener objects, and the WavePlayer 

application is required. Instead, Sonic Animator may be used to 

perform some of these functions. 

• The ability to move multiple AniScape, MultiScape, or 

Space Array objects (sound or listener) in a two-dimensional 

plane (the X,Y plane) by mouse click-and-drag operations. 

• Playing back script files called “Locator Scripts”. These 

files have the extension “.loc” and can be written using a text 

editor. 

• The ability to record Locator scripts from the graphical 

interface, for later playback. 


Operation 

Starting 

↓ To start the Locator tool 

Figure 47 — Locator main interface 

OTHER SIMULATION TOOLS 

The Locator is started from the Windows 2000 start menu. 

1. From the Start Menu select Huron 3.2 

2. Click on the Locator icon. 

The Locator main interface will appear (see Figure 

47). 

Setting a Symbol to Represent a Sound object or Listener object 

↓ To associate a symbol with a 

location channel 

Each symbol on the Locator Control Canvass can be set to 

represent the location of a sound object in two dimensions. By 

associating one of these symbols with the channel of a sound 

object (set in the AniScape, MultiScape , or Space Array 

simulation tools) it is possible to use the Locator Control 

Canvass to move sound or listener objects around the 

Soundscape. 

• Enter the number of the location channel number next to the 

symbol you wish to represent it. 

The symbol on the Locator Canvass will now control the 

location of the sound object associated with the location 

channel number. 


Moving a Listener Object or Sound Object using the Locator Control Panel 

↓ To move a virtual source or 

listener object 

Tip: To ensure the symbol is 

associated with the correct Sound or 

Listener object, watch the main 

interface of the simulation program 

(eg. MultiScape) to see if the 

appropriate X,Y positions are 

updating. 

Locator Scripts 


Once a symbol has been associated with a sound or listener 

object on the Locator interface, it is possible to move them 

about the Locator Canvass. 

• Click and drag on a symbol in the Locator Control Canvass. 

The Sound object will follow the path of the symbol across the 

virtual Soundscape. 

Locator script files are ASCII text files which control the threedimensional 

location of Sound objects and the Listener object. 

This control is achieved by passing messages to applications 

which interpret the location of the sound within the Soundscape 

such as the Sound Field Filter and MultiScape. In addition to 

sending location information, Locator Scripts can be used to 

send MIDI messages, control the WavePlayer, manipulate the 

Convolver, and connect the Huron with a client machine using 

the Socket tool. 

Important Information Concerning the Writing of Locator Scripts: 

• The Locator will flag any syntax errors in commands and 

will prompt the user to for confirmation to continue or abort 

the script. Valid commands which cannot be processed will 

however be ignored by the Locator. 

• Each script command occurs at a point in time determined 

by the previous “Time” command line. 

• Time is measured in 1/10’s of a second from the time the 

start button is pressed. This measurement is not highly 

accurate, as the Locator has been designed for use as a quick 

demonstration and test tool. 

• Commands and their parameters are separated by spaces. 

• Comments may be entered by preceding a line with a “#” 

character. 

• Locator Scripts are named with the *.LOC extension. 


Locator Commands 

Loading and Playing Locator Scripts 

↓ To load a Locator Script 

↓ To run a Locator script 

Pausing and Restarting the Locator Playback 

↓ To pause a Locator Script. 

↓ To restart a Locator Script 

Recording Scripts 


A full list of Locator commands are given in the Command 

Reference subsection at the end of this section. 

Once a Locator script has been written, it may be played back 

through the Locator Control Panel. 

1. Type the name of the *.LOC file into the Browse Text box. 

— or — 

Click Browse … 

2. Then from the file selection menu choose the appropriate 

*.LOC file. 

3. Click OK. 

Once a Locator Script has been loaded it is possible to run the 

script from the Locator control panel. 

• Click Play. 

The Locator Script file will be run, however the position of the 

items will not be updated graphically on the Locator Canvass. 

Locator scripts may be paused during playback. 

• Click Pause. 

The Locator script will halt. 

• Click Pause. 

The Locator Script will restart from the point at which it was 

paused. 

It is possible to record actions performed through the Locator 

Control Canvass into Locator Scripts for later playback. 

Recorded Locator scripts will contain the following 

information: 

• Location. 

• Angle. 

• UpdateVoice. 

• TriggerVoice. 

• ReleaseVoice. 


↓ To record a Locator Script 

Summary 

• SetRoom. 

• SetDirFactor. 


1. Open a Locator script file into which the script will be 

recorded. Messages may be input through MIDI, TCP/IP 

connection, the Locator itself, or the Tracker application. 

2. Click Record. 

3. Move Sound objects across the Locator Control Canvass as 

appropriate, Send TCP/IP messages, or Use the Tracking 

device. 

4. Click Pause to pause the recording. 

5. Click Play to stop recording the script. 

After play has been clicked the recorded script will play back 

for checking. 

• Locator provides a graphical facility for the movement of 

sound objects around a two-dimensional Soundscape. 

• Locator will accept sound movement inputs from MIDI, 

Tracking devices via the tracker application, TCP/IP and 

Locator specific scripts. 

• Locator provides a method for recording and replaying 

Locator scripts. 


Features 

Requirements 

The WavePlayer-2 

The WavePlayer-2 facilitates the playback of audio sound files 

with a Huron system. WavePlayer-2 performs all processing on 

the Huron’s Pentium processor, freeing DSP resources for use 

by other applications. WavePlayer-2 supersedes the original 

WavePlayer application. 

• Plays back up to 24 channels simultaneously. 

• Can mix multiple voices to each output channel. 

• Sample rate conversion and pitch shifting. 

• Sample pre-loading for fast triggering. 

• Supports .wav files. 

• Streams large audio files. 

• Zero Huron DSP card resource requirement. 

WavePlayer-2 requires the following system components: 

• Intel Pentium II or greater system processor 

• Huron ADAT I/O card. 

• RME Hammerfall audio card and accompanying ASIO 

driver software. 

• Three TOSLINK optical audio connector cables. 

• VRack version 3.2 software or newer. 

Please contact Lake Technology for upgrade information if your 

system does not meet these minimum specifications. 


Hardware Connection 

↓ Note: When the Huron is set to 

synchronise to an external clock 

source, it uses the clock 

information that is obtained 

from Huron input 1. 

Driver Installation 

↓ Installing the Hammerfall ASIO 

driver software 

Operation 

Figure 48 – ADAT Connectors 


The Huron ADAT I/O card must be connected to the RME 

Hammerfall card for correct operation with the WavePlayer-2. 

Refer to Figure 1 and make the following audio connections 

with TOSLINK optical cable: 

• Connect RME Output 1 to Huron Input 1 



The WavePlayer-2 requires ASIO driver software to be installed 

for operation with the RME Hammerfall PCI audio card. Huron 

systems shipped with the RME Hammerfall have the 

companion ASIO driver and control panel software preinstalled. 

• If the RME Hammerfall ASIO driver software and control 

panel are not already installed, follow the installation 

instructions contained within the RME Hammerfall 

documentation. 

WavePlayer-2 is primarily controlled by the Socket application 

and SNAP commands. The WavePlayer-2 VRack interface is 

used for system configuration and indication of currently loaded 

wave files to assist with application development. 


Figure 49 — The WavePlayer-2 interface 


The WavePlayer-2 interface displays the following information 

for each wave file: 

Column Description 

id The wave ID 

name The path 

status The status, which can be either Ready (the file is 

successfully loaded but no voices are bound), Bound (one 

or more voices are bound), or a message describing an error 

encountered while loading. 

ch The number of channels in the wave file 

kHz The sample rate of the wave file 

bits The number of bits per sample 

samps The wave file size in samples per channel 

voices Shows information on bound voices in this format: 

(ID)(status){(channels)} 

where: 

- ID is the ID of the bound voice 

- The status is – (if paused), s (if stopped), l if 

looping, or p if playing. 

- channels is a comma-separated list of bound output 

channel numbers. 


Starting 

↓ To access WavePlayer-2 

↓ Note: Unlike most VRack 

applications, the WavePlayer-2 

will never show any input or 

output patchpoints from within 

the Patchbay. 

Setting the System Sample Rate 

↓ Setting the system clock rate 

↓ Note: This setting will only 

affect the RME Hammerfall’s 

clock rate if the RME 

Hammerfall is in master clock 

mode (the default setting). 

System Latency 

↓ Setting the system latency 

Setting Clock Synchronisation 


The WavePlayer-2 must be loaded into the VRack through the 


the System Tools section of this manual). Once loaded, the 

WavePlayer-2 panel will appear in the VRack. 

• Click on the WavePlayer-2 panel in the VRack. 

The WavePlayer-2 interface will appear (see Figure 49). 

The sample rate of the RME Hammerfall’s clock can be set 

from within the WavePlayer-2 configuration interface. 

• Select the sample rate from the sample rate drop box. 

Setting a smaller buffer size will result in a smaller latency (or 

delay) between calling the SNAPWP2PlayVoice command and 

audio being played. Smaller buffer sizes require greater CPU 

load. 

• Refer to the RME Hammerfall documentation for 

instructions for checking its buffer size settings. 

Set the RME Hammerfall card to supply the master clock, and 

set the Huron to synchronise to it. 


↓ Setting the RME Hammerfall as 

the clock master 

↓ Setting the Huron to 

synchronise to the Hammerfall’s 

clock 

↓ Checking clock synchronisation 

↓ Note: If the status indicators are 

yellow, ensure the Huron been 

set to synchronise to an external 

clock. If the status indicators 

are red, check all TOSLINK 

cable connections. 

WavePlayer-2 Audio Outputs in the Huron Patchbay 

Controlling WavePlayer-2 


• Refer to the RME documentation for instructions for setting 

the RME Hammerfall to supply the master clock. 

1. Click on Options/Configure menu in VRack. 

2. Select External. 

3. Click OK. 

1. Click on the IO Manager panel in the VRack interface. 

2. Check the status indicators on input channels 1 – 24. Each 

channel will have a green status indicator if the clocks have 

been synchronised correctly. 

Audio generated by the WavePlayer-2 is accessed via the 

ADAT input channels in the Huron Patchbay. The channel 

mapping is shown below: 

WavePlayer-2 Output Patchbay Input Channel 

Channel 

1 to 8 ADAT Input 1–1 

to ADAT Input 1-8 





It is not possible to directly load wave files into the 

WavePlayer-2 via its VRack interface. Files must be loaded 

and subsequently triggered via SNAP, with the Socket 

application running on the Huron. All audio files must be in 

.WAV format. 

The SNAP commands to run WavePlayer-2 are described in the 

Spatial Network Audio Protocol (SNAP) subsection of this 

manual. 


Optimising System Performance 

↓ Achieving optimum system 

perfomance 

↓ Note: Audio stuttering occurs 

when the system processor is not 

fast enough to keep up with 

audio processing demands. 

Summary 


• Use only .WAV audio files that have the same sample rate as 

the master clock. While WavePlayer-2 can playback audio 

files of other sample rates, CPU intensive calculations are 

required that can greatly reduce system performance. 

• Keep the number of channels that receive pitch-bend 

messages to a minimum. Pitch-bend messages require CPU 

intensive calculations and can reduce system performance. 

• Use the maximum buffer size with the longest system 

latency. This will greatly reduce the load on the processor. 

If the above optimisation tips have been followed and audio 

stuttering still occurs, reduce the number of simultaneous voices 

or upgrade to a faster Pentium processor. 

• WavePlayer-2 allows the loading and playback of .WAV 

format audio files. 

• The WavePlayer-2 is controlled by Socket with SNAP 

commands. 

• The WavePlayer-2 can play up to 24 simultaneous output 

channels using the system processor. 

• Audio generated by the WavePlayer-2 is accessed via the 

ADAT inputs in the Huron Patchbay. 


Note: The Waveplayer has now 

been superseded by Waveplayer-2 

Features 

Operation 

The WavePlayer 

The WavePlayer plays back sampled audio files through a 

Huron system. These audio samples are stored in the DSP 

board’s memory, The WavePlayer can store up to 2 

megasamples of audio data. Each DSP will playback up to 8 

channels or “voices” simultaneously, with the upper limit on 

voices determined by the number of DSP cards in the system. 

The WavePlayer provides: 

• Up to eight voices per DSP. 

• Playback of waves with sample rates up to 48kHz. 

• Facilities for multiple wave files, up to a total of 2097150 

samples. 

The WavePlayer is operated remotely by either SNAP or 

Locator commands. 

Figure 50 — The WavePlayer interface 


Starting 

↓ To access the WavePlayer 

Configuring Voices 

↓ To adjust the number of voices 

played by the WavePlayer 

Loading, Playing, Stopping and Controlling WavePlayer 

wavtosim [-s] [-l] [-S] [-E] 

 

specify s if the wave file already uses the *.SIM 

file format. l specifies the addition of a loop to 

the sample, and S and E specify the start and 

end samples to be converted. 


The WavePlayer must be loaded into the VRack through the 


the System Tools section of this manual). Once loaded, the 

WavePlayer panel will appear in the VRack, and its associated 

patchset will appear in the Patch Bay tool. 

• Click on the WavePlayer panel in the VRack. 

The WavePlayer configuration window will appear (see Figure 

50). 

The WavePlayer supports the simultaneous playback of 

multiple wave files or “voices”. Each DSP in the Huron system 

can play back up to eight simultaneous voices, with the 

WavePlayer being able to control a total of up to two million 

samples of wave data. 

• Select the number of voices required from the drop down 

voices list. 

The WavePlayer will reserve the necessary DSP space required 

for playback and provide patchpoints for each voice in the Patch 

Bay. 

It is not possible to directly load wavetable samples into the 

WavePlayer via its VRack interface. Files must be loaded and 

subsequently triggered via SNAP or Locator commands. All 

wave files must be in a 24bit binary *.SIM format, and must 

have an even number of samples. 

Standard wavefiles may be converted to the *.SIM format by 

using the wavtosim program contained in the 

c:\Huron32\bin directory. 

The two most common methods for loading and triggering 

wave files in the WavePlayer are using a Locator script, or by 

issuing TCP/IP command packets via SNAP and the Socket 

application. The commands to perform these functions are 

described in the SNAP and Command Reference subsection at 

the end of this section. 


Summary 


• WavePlayer allows the loading and playback of wave files in 

24 bit Real .SIM file format. 

• The WavePlayer is controlled with SNAP or Locator 

commands. 

• The WavePlayer can play up to eight voices per DSP card. 

• The WavePlayer allows on the fly updating of voice pitch 

and gain levels. 


Features 

Make Room 

Make Room is a virtual room generation application which runs 

independently of the VRack. It provides room impulse 

response filters for use in the Lake Technology Huron 

environment. Make Room uses statistical simulation to 

generate impulse response files for a huge variety of acoustic 

spaces. 

• Creates filters for both static and dynamic auralisation. 

• Simulates responses through various types of Microphones, 

including Mono Microphones, Blumlein Stereo 

Microphones, and a B-format Microphone. 

• Generates room impulse response filters with user 

configurable sample rates and lengths. 

• Provides the ability to set: sound source distances, azimuth 

and elevations, within variably sized and shaped rooms. 

• Allows control over reverberation times in high and low 

frequencies. 


Operation 

Starting 

↓ To start the Make Room 

application 

↓ To configure a statistical room 

simulation 

Figure 51 — The Make Room interface 

Select Output Mode 


The Make Room application is started from the Windows 

Program Manager. 

1. From the Windows 2000 Start Menu select 

Huron 3.2. 

2. Click the Make Room Icon 

The Make Room interface will be displayed (see Figure 51). 

The Make Room application is parameter based statistical room 

impulse generator. It is necessary to define the properties of the 

room, and the location of the sound object before generating a 

room. Room properties fall under the following categories and 

can be configured using the following steps: 

The output mode specifies how surface reflections present in 

the room impulse response are to be handled. The two output 

mode options are Static and AniScape. 


↓ To select an Output mode 

Select microphone 


d = 0.000 Omni 






• AniScape mode returns a direct sound response with the first 

order (early) reflections missing (this is because AniScape 

provides the first order reflections during a simulation). 

• Static mode returns a sound response file which includes the 

first order reflections of the simulated room, this means that 

the sound object will appear to be “static” in the room (first 

order reflections will always appear to come from the same 

place). 

It is possible to use AniScape mode sound response files with 

MultiScape and Space Array. However, MultiScape and Space 

Array do not add first order reflections during a simulation. 

When producing a room response for a MultiScape or Space 

Array simulation AniScape mode is preferred over the static 

mode, as static impulse responses will not appear to move 

through the Soundscape correctly (ie. the first order reflections 

do not change locations while later reflections do). 

When AniScape mode is selected the Make Room application 

will produce a room geometry file (*.GEO). The *.GEO file is 

a text file containing information on the room’s attributes used 

to calculate first order responses during an AniScape 

simulation. 

• Click Static for Static mode output. 

— or — 

Click AniScape for AniScape mode output. 

There are three types of microphone simulated in the Make 

Room application. Each microphone type mimics the 

properties and produces signal files which mirror the signals 

generated by their real life counterparts. 

Most microphones (with the exception of B-format 

microphones), have characteristic sensitivity lobes. These lobes 

correspond to the sensitivity of the microphones in certain 

directions. In the Make Room application this effect is 

simulated by including a directivity factor for the microphone in 

calculations. 

The Directivity Factor which describes the characteristic sound 

lobes of a Microphone are described in the table below. 

The three types of microphone simulated in Make Room are 

Mono, Stereo and B-format. Each type, their properties and 

resultant responses are described below: 

Mono Microphone: This setting allows the simulation of a 

simple monophonic microphone. The microphone is 


↓ To select a microphone setting 

↓ To specify the file root and 

name of the room response 

Specify the file root and name 


characterised by a directivity factor and can be placed anywhere 

within the simulated room. The room response generated using 

the monophonic microphone is characterised by a single 

impulse response (*.SIM) file. 

Stereo Microphones: This setting simulates a pair of Blumlein 

Microphones. This stereo pair can be imagined as two 

microphones joined at the rear, with an angle of separation 

measured between the pickups of the microphones. It is 

assumed that the microphones are identical and have the same 

directivity factor. The room response generated using the stereo 

pair is characterised by two impulse files named *L.SIM, 

*R.SIM respectively. 

B-format Microphone: This setting simulates a B-format 

microphone which records four separate signals. These signals 

represent the X,Y,Z and Omni (ω) encoding of the room 

impulse. Combined, these signals provide responses which 

allow clear sound localisation using Ambisonics. The room 

responses generated using the B-format setting are characterised 

by four impulse files named *X.SIM, *Y.SIM, *Z.SIM and 

*W.SIM respectively. 

• Click Mono Microphone for monophonic microphone 

output. 

—or — 

Click Blumlein Stereo for Stereo microphone output. 

—or — 

Click B-format for B-format microphone output. 

The file root and name specify the directory to which impulse 

responses are saved and the name of the impulse response files 

(*.SIM) produced by the simulation. 

For example, a file root c:\Huron32\rooms 

Would save the impulse responses in the directory 

c:\Huron32\rooms, the sim files would be named 

ROOM*.SIM, where * represents any special file designators 

(ie. R,L,X,Y,Z,W). 

If AniScape mode is selected, Make Room will also generate a 

geometry file (*.GEO) in the file root directory. 

1. Click Browse. 

2. Type or Select the output directory and specify a name for 

the output files. 

3. Click OK. 

Specify sample rate and length in samples 

The sample rate specifies the number of samples of the room 

response to be taken during one second. AniScape and 

MultiScape use sampling rates of 16000 due to band limiting 

within the programs, while most other Huron applications 

(including Space Array) utilise a sampling rate of 48000. These 


↓ To set the sample rate 

↓ To set the length of the impulse 

↓ To specify the Room Size 

↓ To specify the Room Shape 

↓ To place a sound source 

Specify room size and a room shape 

Placing a sound source 

Specify Echo settings 


“standard” rates may change if the external clock feature of the 

VRack interface is being used. 

The length in samples specifies the length of the impulse 

response of the room. The minimum number of samples 

required to capture a whole room response can be determined 

by multiplying the reverberation time (in seconds) by the 

sample rate. 

• Type the number of samples per second (Hz) in the Sample 

Rate Box. 

1. Determine the minimum sample length by multiplying total 

reverberation time by the sample rate. 

2. Type the total sample length in samples in the Length Box. 

The room response is statistically generated using two main 

variables, The room size and the room shape. 

Room Size is described using the Mean Free Path. The Mean 

Free Path is a statistical measure stated as “the average (mean) 

distance a sound originating at one wall has to travel to reach 

another wall in meters”. 

The Room Shape is a random number seed, used during 

calculations to provide a room shape. Due to the nature of the 

algorithm that Make Room uses, a specific seed will 

consistently produce the same sounding room response, these 

room responses are fairly constant even as the room size 

changes. 

• Type a number in meters for the size of the room in the 

Room Size box. 

• Type a number between 0 and 1 in the Room Shape box. 

Placing a sound source involves specifying the location of the 

sound source in relation to the microphone. The microphone is 

always assumed to be at the centre of the room, thus distances 

and the location of the sound source are given in meters and 

degrees from this central point. The values required are the 

distance from the microphone in meters, the azimuth of the 

sound source in degrees and the elevation of the sound source in 

degrees. 

1. Type a value in meters in the Mic. Distance box. 

2. Type a value in degrees in the Src Azimuth box. 

3. Type a value in degrees in the Src Elevation box. 

Two Echo settings must be specified, 


↓ To specify the Echo settings 

↓ To set the reverberation time 

for Low frequencies 

↓ To set the reverberation time 

for High frequencies 

Define reverberation time 

Define Diffusion time 

↓ To set the Diffusion of sound at 

low frequencies 

↓ To set the Diffusion of sound at 

high frequencies 

Generate impulse signals 


• Echo Delay is the time in seconds that the first order 

reverberation / echoes take to reach the microphone. This 

value is ignored in AniScape mode, as no first order 

reflections are produced. 

• Echo elevation is the elevation from which the echoes 

appear to originate. For example. An elevation of -20 

degrees simulates reflections coming off the floor of a room, 

while an elevation of 45 degrees simulates reflections 

coming from the ceiling. 

1. Type the time for echo delay in seconds in the Echo Delay 

box. 

2. Type the echo elevation in degrees in the Echo Elevation 

box. 

Reverberation time represents the amount of time taken for the 

sound pressure of an initial sound source to drop by 60db. For 

example, a handclap with an initial sound pressure level of 

30dB in a room with a reverberation time of 2.00 seconds, will 

take 2.00 seconds to reach a volume of -30dB. Setting longer 

reverberation times, produces more echoic room responses. 

Reverberation times are added to the room simulation in two 

segments, Low frequency, and High frequency. 

• Type the new value in seconds into the low freq Reverb 

Time box. 

• Type the new value in seconds into the high freq Reverb 

Time box. 

Diffusion is the amount of atrophy that occurs to a sound signal 

after it reflects off a room surface. For example, if the diffusion 

amount is set at 10.00% the first time a sound hits the wall only 

90% of the existing sound will be reflected. 

• Type the new value as a percentage loss in the low freq 

Diffusion box. 

• Type the new value as a percentage loss in the high freq 

Diffusion box. 

Once all the values for the room have been set it is possible to 

generate the room impulse signals. 


↓ To generate the impulse signals 

for the room 

Saving a Make Room Setup 

↓ To save a Make Room setup 

Restoring a Make Room Setup 

↓ To restore a previously saved 

Make Room file 

Summary 


• Click Generate. 

A status update will appear on the Generate button and once 

complete, the impulse responses (*.SIM) and associated 

geometry file (*.GEO) will be saved in the File Root directory. 

Make room setups may be saved so that they may be used to 

generate the same room response at a later date. 

1. Click Save As. 

A Save As dialogue box will appear. 

2. Type the name of the file to save. 

—or — 

Select an existing file name to save over. 

3. Click OK to save the *.WSP. 

1. Click Restore. 

2. Select the file you wish to restore from the dialogue. 

3. Click OK. 

• Make Room provides a facility for statistically generating 

room responses from a series of parameters. 

• Make Room simulates three types of microphone pickups in 

simulated rooms, generating appropriate *.SIM *.GEO files 

for each type. 

• Make Room generates signals for both AniScape and Static 

sound simulations. The AniScape signals may be used in 

MultiScape and Space Array. 


Integration under 32 bit Windows 

Integration under Unix 

API 

Spatial Network Audio Protocol 

(SNAP) 

The Spatial Network Audio Protocol (SNAP) allows audio 

software developers to communicate with a Huron workstation 

via a TCP/IP network. It simplifies the task of creating and 

sending LakeNet messages to the Huron by encompassing the 

valid LakeNet messages in a distributable module. This module 

has a common applications programmer interface (API) for both 

Win32 and Unix platforms. 

SNAP is distributed as source code and a static library to allow 

easy integration with new applications under 32 bit Windows 

systems or the various Unix platforms. 

SNAP consists of the files SNAP.h and SNAP.c. For the 

Windows implementation, a library file (SNAP.lib) is also 

included. When linking a project that uses SNAP, link to 

SNAP.lib and the Windows Sockets library ws2_32.lib. 

ws2_32.dll must also be available on the system path. 

SNAP consists of the files SNAP.h and SNAP.c. Compile and 

link SNAP.c with your application. You must defining the 

symbol unix. Example: 

cc SNAPTest.c SNAP.c -Dunix 

The API is specified fully in the header file SNAP.h. An 

explanation of the parameters for each function is given below. 

The convention for input and output parameters is that inputs 

are basic types and outputs are pointers. The exception is 

strings, where inputs are declared const and outputs have no 

such definition. 


Initialisation 

SNAP Commands 

General VRack Commands 

Parameter Description 


To begin communication, issue the SNAPConnect() command. 

This will open a socket to the Huron. Note that the Socket for 

Windows 2000 application must be running on the Huron or 

this command will fail. Use SNAPDisconnect() to close the 

socket. If your application exits without calling 

SNAPDisconnect() then Socket for Windows 2000 may crash. 

SNAPConnect 

Creates a TCP/IP connection with the Huron. The Socket 

application must be running on the Huron for a connection to be 

established. 

in const char* szIPAddress The IP address of the Huron 

SNAPDisconnect 

Disconnects the TCP/IP connection with the Huron. 

SNAPExec 

Huron runs a command on the Microsoft Windows operating 

system. To send a command to be executed by Windows call 

this function with the full path of the command in the path field. 

in char* szPath Full path of Windows command 

SNAPSetWorkspace 

This command allows the loading of a new workspace into the 

VRack of Huron. This command requires the path field to be 

set to the full path of the workspace file. VRack must also be 

running on the Server (Huron). 

in const char* szPath Full path of workspace file 

out long* plStatus 

out char* szPath 

SNAPGetStatus 

This command returns the current operating status of the Huron. 

SNAPGetWorkspace 

To find the currently loaded workspace in VRack send this 

command. The VRack application will send a 

HuronGetWorkspace command back to the remote computer 

with the path field set to the full path of the current workspace 

file. VRack must be running. 


Positional Commands 

Note: The number of positional 

commands sent per second should 

be limited by the controlling 

application. Sending too many 

messages per second can cause 

buffer overflow, resulting in 

delayed response and loss of 

command data. 



These positional commands can be used with Binscape, Space 

Array, Multiscape, Aniscape, and Headscape. The sound 

objects and listener object in each simulation application 

receive their location and orientation information on Location 

and Angle channels. Every independently moveable source 

should have a unique Location and Angle channel assigned to 

it. Object location and angle settings will be updated on receipt 

of the command. 

SNAPLocation 

This command sets the X, Y, and Z coordinates of an object. 

The distances are measured in meters. 

in long lChannel Location channel that will receive the coordinates (1-255) 

in float fX X coordinate of the object in meters 

in float fY Y coordinate of the object in meters 

in float fZ Z coordinate of the object in meters 

SNAPAngle 

This command sets the orientation of an object. The angles 

passed are in degrees. 

in long lChannel Angle channel that will receive the data (1-255) 

in float fAz Azimuth of the object in degrees 

in float fEl Elevation of the object in degrees 

in float fRl Roll of the object in degrees 


AniScape Commands 


c:\Huron32\AniScape\rooms\room.geo 

c:\Huron32\AniScape\rooms\roomx.sim 

c:\Huron32\AniScape\rooms\roomy.sim 

c:\Huron32\AniScape\rooms\roomz.sim 


SNAPLoadRoom 

The Sound Field Filter can be loaded with a up to fifteen 

different room acoustic definitions. Each room is stored in a 

separate buffer. The data for the room acoustic definitions are 

stored in a number of binary data files on Huron, the set of data 

files for a room’s acoustics all have the same file name root but 

different extensions. The path passed in this command should 

be the full path of the data files without the file extension. 

For example, if the set of data files for a room resides in the 

following directory: 

The string passed would be 

c:\Huron32\AniScape\rooms\room in the path field of 

this command. 

This command returns a HuronAck packet to the remote 

computer. The cmd field of the HuronAck packet will contain 

HuronAniLoadRoom and the data field will contain the buffer 

number that the room acoustic files were loaded into. If the 

Sound Field Filter was unable to load the data files it will return 

-1 in the data field of the HuronAck packet. 

This buffer may also be used to switch between and clear rooms 

using the HuronAniSetRoom and HuronAniClearRoom 

commands. 

in const char* szPath The directory path of the data files 

SNAPSetRoom 

The Sound Field Filter can be loaded with a number of different 

room acoustic definitions. Each room is assigned a buffer 

number. This command can be used to switch the Sound Field 

Filter’s room acoustic definition buffer number. 

in long lBuffer The buffer number of the new room 

SNAPClearRoom 

The Sound Field Filter can be loaded with a number of different 

room acoustic definitions. Each room is assigned a buffer 

number. This command can be sent to remove the acoustics 

loaded in the defined buffer number. 

in long lBuffer The buffer number of the room to remove 



SNAPGetRooms 

The Sound Field Filter application must be running in the 

VRack to receive a return from this command. 

out long *pnRooms Number of rooms loaded 

SNAPGetRoom 

It is important to send the command with the index field set the 

room you are interested in and the returned packet will contain 

the buffer and the path of the room. In order for this command 

to return the Sound Field Filter application must be running in 

the VRack. 

in long lIndex 0 based index of the room you want information on 

out char* szPath The directory path of the room’s data files 

SNAPSetMinDistance 

The Sound Field Filter has a setting called Minimum Source 

Distance which is measured in meters. This specifies a sphere 

around the listeners position with a radius of the minimum 

source distance value. If a sound object comes within this 

circle, its volume will remain fixed at the level on the edge of 

circle and its directionality will gradually turn to omnidirectional 

as it approaches the listeners position. The 

minimum source distance can be set using this command. 

in float fMSD The minimum source distance in meters 

SNAPGetMinDistance 

In order to receive a return from this command, the Sound Field 

Filter application must be running in the VRack. 

out float* pfMSD The minimum distance in meters 

SNAPSetDirFactor 

Each sound object in the Sound Field Filter can be assigned a 

direction factor, which is a floating point number between 0 and 

1 which describes the radiation pattern for a sound object. 

Directivity Factor Description 

0 Omni directional 

0.5 Cardioid 

1.0 Figure eight 

The source parameter identifies the Sound object and is 

equivalent to the primary location channel. Problems may arise 

if two sources sharing the same location channel have different 

directivity factors. Lake Technology recommends that sound 

objects with different directivity factors should always be 

assigned unique location channels. 

in long fSource The source. 0 is the receiver and the sound objects are 

numbered 1,2,3…etc in the order they appear in the Sound 

Field Filter 

in float fDirFactor The direction factor 



SNAPGetSources 

To receive a reply from this command the Sound Field Filter 

application must be running in the VRack. 

out long *plSources Number of sound objects currently active in Sound Field Filter 

in long lIndex 0 based index of the source you want information on. (0 is 

always the receiver). 

out long* plLoc1 The primary location channel used by this source 

out long* plLoc2 The secondary location channel used by this source 

out long* plAng1 The primary angle channel used by this source 

out long* plAng2 The secondary angle channel used by this source 

out float* pfDirFactor The direction factor of this source (not applicable to the 

listener). 

SNAPWaveDoppler 

Enables or disables doppler shifting of waves in the 

WavePlayer via the Sound Field Filter. If enabled AniScape is 

deals with the doppler shifting of waves. If disabled then the 

driving application is responsible for providing doppler shifting. 

in long lOn Doppler shifting Enabled or Disabled 

SNAPSetAttenuation 

Enables or disables the attenuation of sound objects due to 

distance by the Sound Field Filter. If disabled the driving 

application should provide range attenuation. 

in long lOn Distance attenuation enabled or disabled 


MultiScape Commands 



SNAPSetAcousticSpace 

The SNAPSetAcousticSpace command assigns a number and 

loads the filters for an acoustic space into MultiScape. 

in long lSpace set acoustic space 

in const char* szPath directory path of the room’s data files 

SNAPMultiSetRoom 

The SNAPMultiSetRoom command associates a room number 

with an acoustic space. 

in long lSpace The identifier of the acoustic space 

in long lRoom The room number to associate with the acoustic space 

SNAPMultiSetRoomSegment 

The SNAPMultiSetRoomSegment sets the geometry for a room. 

The first parameter is the room number followed by the X,Y,Z 

of the bottom left corner and the X,Y,Z of the top right corner 

of the room segment. A room can be made up of many room 

segments. 

in long lRoom The room number 

in float fmin_xu Float representing the minimum X coordinate of the a room 

segment (bottom left). 

in float fmin_yu Float representing the minimum Y coordinate of the a room 


in float fmin_zu Float representing the minimum Z coordinate of the a room 


in float fmax_xu Float representing the maximum X coordinate of the a room 

segment (top right). 

in float fmax_yu Float representing the maximum X coordinate of the a room 


in float fmax_zu Float representing the maximum X coordinate of the a room 




SNAPMultiSetDoor 

SNAPMultiSetDoor sets the coordinates of the door, and the 

rooms it connects. A door must have has a width of 0. 

in long lDoor The door identification number 

in long lRoom1 The first room identification number to which the door 

connects 

in long lRoom2 The second room identification number to which the door 

connects. 

in float fmin_xu Float representing the minimum X coordinate of the door 

(bottom left). 

in float fmin_yu Float representing the minimum Y coordinate of the door 


in float fmin_zu Float representing the minimum Z coordinate of the door 


in float fmax_xu Float representing the maximum X coordinate of the door (top 

right). 

in float fmax_yu Float representing the maximum X coordinate of the door (top 

right). 

in float fmax_zu Float representing the maximum X coordinate of the door(top 

right). 

SNAPMultiOpenDoor 

Doors can be opened or closed. A fully open door has a value 

of 1.00, a closed door has a value of 0.00. 

in long lDoor door identification number 

in float fOpen Float representing the state of the door. 

SNAPMultiResetMap 

Before Sending information about a new simulation to 

MultiScape (e.g new acoustic spaces or room segments) the 

driving application should send a ResetMap to clear out any old 

information since last time MultiScape was run. 


Space Array Commands 



Space Array accepts Location and Angle commands in the same 

way as AniScape and MultiScape. It also recognises the 

AniScape room switching commands. 

SNAPLocation 

This command sets the X, Y, and Z coordinates of an object. 

The distances are measured in meters. 

in long lChannel Location channel that will receive the coordinates (1-255) 

in float fX X coordinate of the object in meters 

in float fY Y coordinate of the object in meters 

in float fZ Z coordinate of the object in meters 

SNAPAngle 

This command sets the orientation of an object. The angles 

passed are in degrees. 

in long lChannel Angle channel that will receive the data (1-255) 

in float fAz Azimuth of the object in degrees 

in float fEl Elevation of the object in degrees 

in float fRl Roll of the object in degrees 

c:\Huron32\SpaceArray\rooms\room.geo 

c:\Huron32\SpaceArray\rooms\roomx.sim 

c:\Huron32\SpaceArray\rooms\roomy.sim 

c:\Huron32\SpaceArray\rooms\roomz.sim 

SNAPLoadRoom 

The Space Array can be loaded with up to two different room 

acoustic definitions. Each room is stored in a separate buffer. 

The data for the room acoustic definitions are stored in a 

number of binary data files on Huron, the set of data files for a 

room’s acoustics all have the same file name root but different 

extensions. The path passed in this command should be the full 

path of the data files without the file extension. 

For example, if the set of data files for a room resides in the 

following directory: 

The string passed would be 

c:\Huron32\SpaceArray\rooms\room in the path field 

of this command. 

SNAPLoadRoom will load the room into the current buffer, 

which defaults to buffer 0 when Space Array is loaded. To load 

a room into the second buffer, first issue the SNAPSetRoom 

command with the parameter 1 to switch to the second buffer 

and then load the desired room. 



This command returns a HuronAck packet to the remote 

computer. The cmd field of the HuronAck packet will contain 

SNAPLoadRoom and the data field will contain the 

buffernumber that the room acoustic files were loaded into. If 

the Space Array was unable to load the data files it will return - 

1 in the data field of the HuronAck packet. 

This buffer may also be used to switch between and clear rooms 

using the SNAPSetRoom and SNAPClearRoom commands. 

in const char* szPath The directory path of the data files 

SNAPSetRoom 

The Space Array can be loaded with a number of different room 

acoustic definitions. Each room is assigned a buffer number. 

This command can be used to switch the Space Array’s room 

acoustic definition buffer number. 

SNAPSetRoom must be issued before loading a room into any 

buffer but the default one (buffer 0). 

in long lBuffer The buffer number of the new room 

SNAPClearRoom 

The Space Array can be loaded with a number of different room 

acoustic definitions. Each room is assigned a buffer number. 

This command can be sent to remove the acoustics loaded in the 

defined buffer number. 

in long lBuffer The buffer number of the room to remove 

SNAPSetMinDistance 

The Space Array has a setting called Minimum Source Distance 

that is measured in meters. This specifies a sphere around the 

listener’s position with a radius of the minimum source distance 

value. If a sound object comes within this circle, its volume 

will remain fixed at the level on the edge of circle and its 

directionality will gradually turn to omni-directional as it 

approaches the listener’s position. The minimum source 

distance can be set using this command. 

in float fMSD The minimum source distance in meters 


VMixer Commands 



SNAPVMxLoad 

Instructs VMixer to load a .vmx configuration file from disk. 

in const char szPath[128] Path of the .vmx file to load 

SNAPVMxSave 

Instructs VMixer to save its current settings to a .vmx 

configuration file. 

in const char szPath[128] Path of a .vmx file to save the current configuration to 

SNAPVMxSetSize 

Sets the number of inputs and outputs of the VMixer. 

in long lInputs New number of inputs 

in long lOutputs New number of outputs 

SNAPVMxGetSize 

When a VMixer receives this command, it fills the fields with 

the current number of inputs and outputs. 

out long* plInputs Current number of inputs 

out long* plOutputs Current number of outputs 

SNAPVMxSetNodeGain 

Sets the gain of a matrix node to the given gain value. 

in long lInputChannel Input channel of the node 

in long lOutputChannel Output channel of the node 

in float fGain The new gain value between 0.0 and 4.0 

SNAPVMxSetInputGain 

Sets the preamplification gain of a mixer input. This gain is 

applied before the matrix gain and the output gain. 

in long lInputChannel The number of the input channel (0-based) 


SNAPVMxSetOutputGain 

Sets the postamplification gain of a mixer output. This gain is 

applied after the input gain and the matrix gain. 

in long lOutputChannel The number of the output channel (0-based) 


SNAPVMxGetNodeGain 

When the VMixer receives this command, it fills the gain field 

with the current gain of the requested node. 

in long lInputChannel The input channel identifying the node 

in long lOutputChannel The output channel identifying the node 

out float* pfGain The current gain at this node 



SNAPVMxGetInputGain 


with the preamplification of the requested input channel. 

in long lInputChannel The requested input channel 

out float* pfGain The current preamplification on this channel 

SNAPVMxGetOutputGain 


with the preamplification of the requested input channel. 

in long lOutputChannel The requested output channel 

out float* pfGain The current postamplification on this channel 


WavePlayer-2 Commands 


The WavePlayer-2 is a Huron application that can play .WAV 

format audio files. WavePlayer-2 does not use any Huron DSP 

resources, and it is not constrained to the 2Mb memory limit of 

the original WavePlayer application. 

The SNAPWP2LoadWave function blocks and returns only 

after the load has succeeded or failed. The return code reflects 

whether WavePlayer-2 was able to successfully load the wave 

file. All other WavePlayer-2 functions are non-blocking. 

Their return codes indicate whether each command was 

successfully sent to the Huron. 

Valid wave IDs are 1 to 256 inclusive. Valid voice IDs are 1 to 

128 inclusive. 

The sample code shown below provides an example of how to 

control WavePlayer-2 using the SNAP API. 

if (SNAPConnect("127.0.0.1")) 

{ 

/* Load sample.wav as wave 1 */ 

if (!SNAPWP2LoadWave(1, "C:\\wavs\\sample.wav")) 

printf("Could not load wave file.\n"); 

else 

{ 

/* Bind voice 5 to wave 1, outputting on channels 3 and 4 */ 

long lChans[2] = {3, 4}; 

SNAPWP2BindVoice(1, 5, 2, lChans); 

} 

/* Start voice 5 playing */ 

SNAPWP2PlayVoice(5); 

/* Wait for voice to finish playing */ 

... 

/* Release the voice */ 

SNAPWP2ReleaseVoice(5); 

/* Remove the wave */ 

SNAPWP2RemoveWave(1); 

} 

SNAPDisconnect(); 




SNAPWP2LoadWave 

Loads a wave file into the WavePlayer-2 and assigns it an ID. 

Standard .wav file formats are supported including mono and 

multi-channel. At least one voice must be bound to the wave 

before playback can occur. For optimum performance, it is 

recommended that wave files be selected that are at the same 

sample rate as the Huron. 

FALSE is returned if the load fails and the reason for failure is 

displayed in the WavePlayer-2 window. 

in long lWaveID A wave ID 

in const char* szPath Full path to the wave file 

SNAPWP2BindVoice 

Binds a voice to a wave ID. A voice is an instance of a loaded 

wave file for playback, allowing polyphony without having to 

load the same wave file multiple times. When a voice is bound 

it is stopped, unpaused, not looped and has a gain of 1 and a 

pitchbend factor of 1. 


in long lVoiceID A voice ID 

in long lNumChans The number of channels to be bound to this voice. This must 

be less than or equal to the number of channels in the wave 

file. 

in long* plChans An array of length lNumChans containing 1-based output 

channel numbers mapping each wave file channel to an output 

channel on the sound card. Multiple voices can mix into the 

same output channel. 

SNAPWP2UpdateVoice 

Updates a voice’s level and pitch. This can be done either 

before or during playback. 

In long lVoiceID A voice ID 

in float fSPL Gain of this sound 

in float fPBend Pitchbend factor for this sound. 0.5 is half pitch, 1.0 is normal 

pitch and 2.0 is double pitch 

SNAPWP2PlayVoice 

Starts a voice playing from the beginning of the wave file. If it 

is already playing, restarts it from the beginning. 


SNAPWP2LoopVoice 

Sets a voice’s loop state. 


in long lStartSamp The sample number at which the looped section begins 

in long lEndSamp The sample number up to which the section is looped. If 

lEndSamp is set to 0, the wave file is looped to the end. 



SNAPWP2PauseVoice 

Pauses a voice. Voices can be paused before they are played, in 

which case they must be unpaused before playback occurs. 

SNAPWP2PlayVoice does not unpause a paused voice. 


SNAPWP2UnpauseVoice 

Unpauses a previously paused voice. 


SNAPWP2StopVoice 

Stops a playing voice. 


SNAPWP2ReleaseVoice 

Performs the inverse of SNAPWP2BindVoice. Once a voice is 

released, all other commands for that voice ID are ignored 

(except SNAPWP2BindVoice). 


SNAPWP2RemoveWave 

Removes a wave file from WavePlayer-2, automatically 

stopping and releasing any bound voices. 


SNAPWP2RemoveAllWaves 

Removes all wave files from WavePlayer-2, automatically 

stopping and releasing any bound voices. 


WavePlayer Commands 



The WavePlayer is a Huron application that can play 24 bit 

audio data in .Sim file format only. Data is played out of Huron 

DSP memory. Multiple voices can be played at varying pitches 

and amplitudes. It can load a large number of different audio 

waves and play any wave on any voice. Up to 2 Mb of audio 

data can be loaded into the WavePlayer. 

When wave data is downloaded to the Huron from the remote 

computer a wave name and checksum must be passed in the 

SNAPSetWaveHeader command, followed by the data in the 

SNAPSetWaveSamples command. When 

SNAPSaveTransferWave is issued the data is saved onto the 

Server (Huron’s) hard disk drive and downloaded into the 

WavePlayer. 

The SNAPLoadWave command requires that a wave name and 

a checksum is sent. The Huron checks these parameters against 

the waves stored on its hard disk drive, and will load the wave 

directly from its own disk, if it can. 

The WavePlayer plays voices. To attach a wave to a voice use 

the SNAPUpdateVoice command, and then use 

SNAPTriggerVoice and SNAPReleaseVoice to play and stop 

the voice playing. While the voice is playing it is possible to 

send further SNAPUpdateVoice commands to change the pitch 

and the level of the voice. 

SNAPGetWaveProp 

This command is received by the WavePlayer application which 

sends a SNAPGetWaveProp packet back to the remote 

computer with the variables of the SNAPGetWaveProp 

structure filled in as follows. The WavePlayer application must 

be running in the VRack for this command to return. 

out long* plVersion Version of WavePlayer TCP/IP Interface 

out long* plFlag Currently unused 

out long* plVoices Number of voices currently selected in the WavePlayer 

out long* plWaves Maximum number of waves that can be loaded 

out long* plTotalSamples Maximum number of 16 bit samples that Huron can store 

out long* plFreeSamples Number of free samples not used for storage 



SNAPSetWaveHeader 

When downloading fresh wave data to the Huron from the 

remote computer this command must be sent prior to the 

sending the data for the wave. 

in long lWaveID The waveID that you want to assign to this wave. If the 

waveID is already taken by another wave, that wave has the ID 

removed from it. A single wave can be assigned multiple 

waveIDs. It is removed from the WavePlayer’s memory when 

its last ID is removed. 

in long lFrames Number of frames in the waveform. A frame contains a 

sample for each channel. 

in long lSRate Sample rate of the wave. 

in long lLoopType Currently ignored. If loopStart is less than loopEnd then the 

wave will loop forward, otherwise the wave will only play 

once when triggered 

in long lLoopStart Start sample of loop 

in long lLoopEnd End sample of loop 

in const char* szWaveName Name of the wave 

in long lChecksum Checksum 

SNAPSetWaveSamples 

Once the header has been sent the data can be sent using the 

SNAPSetWaveSamples command. 

in long lWaveID The waveID that you sent in the SNAPSetWaveHeader 

command 

in long lChannel Channel number. As only mono waves are supported this field 

is ignored. 

in long lOffset Offset in samples from the start of the waveform that this data 

is for. 

in long lLength Number of samples that is being set. 

in short* psSamples The data. Only HURONTCP_MAX_SAMPLES (defined in 

HuronTCP.h) may be transferred with one command. If your 

wave is bigger then you must split the wave into chunks and 

send it piece by piece. 

SNAPSaveTransferWave 

Once all the data has been sent to Huron the 

SNAPSaveTransferWave command must be sent. This 

command will cause the data to be saved to Huron’s disk and 

transferred to the WavePlayer’s memory. 

in long lWaveID The waveID that you sent in the SNAPSetWaveHeader 

command 

SNAPLoadWave 

If the wave required is already stored locally on the Huron, this 

command may be sent. The Huron will attempt to find the 

wave at the specified path. The Huron stores wave data in SIM 

files. See the SIM Tools section in this manual for an 

explanation of the SIM file format. 

If the load message is not received, a -1 is returned in the data 

field. 



in long lWaveID The waveID that you want to assign to this wave. If the 

waveID is already taken by another wave, that wave has the ID 

removed from it. A single wave can be assigned multiple 

waveIDs. It is removed from the WavePlayer’s memory when 

its last ID is removed. 

in const char* szPath Full path of a SIM file 

SNAPUpdateVoice 

After the waves have been loaded it is necessary to assign them 

to a voice in order to play the wave. To do this send the 

SNAPUpdateVoice command. Once the wave is playing it is 

possible to send more SNAPUpdateVoice commands to change 

the pitch and level of the voice. 

in long lVoice Voice number 

in long lWaveID WaveID to assign to this voice 

in float fSPL Sound pressure level of this voice between 0 and 1 

in float fPBend Pitchbend value of this voice. 0 turns the voice off, 0.5 is half 

pitch, 1.0 is normal pitch and 2.0 is double pitch 

SNAPTriggerVoice 

Starts a voice playing. 


SNAPReleaseVoice 

Stops a voice playing. 


SNAPRemoveWave 

Allows the driving application to remove a wave from the 

WavePlayer. 

in long lWaveID The identification number of the wave file to remove 

SNAPGetWaveInfo 

After sending a SNAPGetWaveProp the driving application is 

returned the number of loaded “waves” parameter of the 

returning structure. After this return is made, the 

SNAPGetWaveInfo will return information on all these waves 

by passing the “index” which is a number from 0 to “waves” 

(derived from SNAPGetWaveProp). 

If the wave was loaded by the driving application the structure 

returned provides the lWaveID and its complete path in “file” or 

its “szWaveName” and “plChecksum”. 

out long* lWaveID Identification number of the wave file 

in long lIndex The WavePlayer voice number 

out char* szPath Full path of a SIM file 

out char* szWaveName Name of the wave 

out long* plChecksum Checksum 


Command Reference 

This section describes commands that can be used for 

controlling the Huron. 

Remote control of the Huron is normally performed using 

SNAP and the Socket application. SNAP supersedes the Huron 

LakeNet interface for sending Huron control messages via 

TCP/IP. Please refer to the SNAP section for further 

information on SNAP commands. LakeNet documentation can 

be found in PDF format on the Huron VRack CDROM. 

For control of Huron applications using Windows Messaging, 

please refer to the LakeBus documentation that can be found in 

PDF format on the Huron VRack CDROM. 

This section provides information about scripting commands 

that can be used by Locator and Sonic Animator. These 

programs parse a script and translate ASCII commands into 

commands that can be interpreted by the Huron. Both 

applications can be run locally on the Huron, or on a remote 

machine when used in combination with the Socket application. 


Locator Commands 

General Commands 

Source and Listener Object Control 


This subsection lists the commands recognised by Locator when 

parsing Locator scripts. 

Time T 

All following commands up to the next Time command will be 

executed T (Integer +ve) tenths of a second after the Play button 

is clicked. 

SetWorkspace workspace 

Set VRack workspace to “workspace” (String as a path to a file 

and filename with a *.WSP extension). 

Exec command 

Execute command (String) in Windows shell. If the exec 

command is launching the VRack, the exec command requires 

that the full path to the *.WSP file is given eg. 

c:\huron30\rack32.exe c:\huron30\bin\huron.wsp 

Repeat 

Set time index to 0 - Restarts the script. 

# 

Comment line. All chars to the end of the line are ignored. 

Loc N X Y Z 

Move the object specified by location channel N (Integer +ve) 

from its current position, to the location X,Y,Z (Float) 

coordinates (in metres from origin). Location channel numbers 

must be in the range 1 to 255. 

Ang N A E R 

Send Angle message with Azimuth A (0-360 degrees), 

Elevation E (0-360 degrees) and Roll R (0-360 degrees) degrees 

on Angle channel N (Integer +ve). 

Jump N X Y Z vX vY vZ 

Set object at location channel N (Integer +ve) to new location 

X,Y,Z ( in meters from origin) and set velocity in the X plane to 

vX, the velocity in the Y plane to vY and the velocity in the Z 

plane to vZ. 

SetDirFactor nSource DirFactor 

(Sound Field Filter) 

This command sets the Directivity Factor, DirFactor (Float) of a 

sound nSource object (Integer, as an index to the objects in the 

sound object list) in Sound Field Filter. 


WavePlayer Commands 


SetDirFactor nSource DirFactor 

(MultiScape) 

This command sets the Directivity Factor DirFactor (Float) of a 

sound object nSource (Integer +ve, Location channel of 

object) in MultiScape. 

LoadWave N simfile.sim 

Loads Wave File name “simfile.sim” (String, Path to wave file 

in *.SIM Format) and assigns a Wave ID of N (Integer +ve). 

UpdateVoice V N G S 0 

This assigns voice V (Integer, 0 to number of voices - 1) of the 

WavePlayer to Wave ID N (Integer +ve), giving it a gain of G 

(Float from 0 to 1), and playing it at speed S (Float 0 to 2, with 

1 being normal). The 0 (Zero) at the end is required but is not 

currently used. 

TriggerVoice N 

Start voice V (Integer, 0 to number of voices - 1) of the 

WavePlayer playing. 

ReleaseVoice N 

Stop voice V (Integer, 0 to number of voices - 1) of the 

WavePlayer. 

RemoveWave N 

Remove wave with Wave ID N (Integer +ve) from the 

WavePlayer. 

Sound Field Filter (AniScape) Commands 

MultiScape Commands 

AniLoadRoom Path 

Load a room specified by the full Path (String, the path 

containing the *.GEO file and associated impulses) into the 

Room dialogue box of the Sound Field Filter. 

AniSetRoom N 

Set Sound Field Filter Room to Buffer N (Integer, 0 to number 

of buffers - 1). 

AniClearRoom N 

Remove a Room specified by Buffer N (Integer, 0 to number of 

buffers - 1) from the Sound Field Filter Dialogue box. When a 

room with a lower index than buffers-1 is removed, the higher 

numbered rooms are shuffled downwards in the list. 

SetAcousticSpace S Path 

Assign an acoustic space specified by the full Path (String, the 

path containing the *.GEO file and associated impulses) to the 

Acoustic Space Identifier S (Integer +ve). 


Note: All doors should have at least 

one plane along which their width 

equals zero (0.00). 

Midi Commands 

Convolver Commands 


SetRoom R S 

Assign an Acoustic Space S (Integer +ve) to a Room R (Integer 

+ve). 

SetRoomSegment R Xmin Ymin Zmin Xmax Ymax Zmax 

Sets the Dimensions for a room segment (Float, XYZmin 

specifies the lower left corner, XYZmax specifies the Upper 

right corner) and associates the segment with Room R (Integer 

+ve). 

SetDoor D R1 R2 Xmin Ymin Zmin Xmax Ymax Zmax 

Creates a Door D (Integer +ve) between Room 1 R1(Integer 

+ve) and Room 2 R2 (Integer +ve), with the dimensions 

specified by XYZ min (Floats, specifies the lower left corner 

coordinates of the door) and XYZ max (Floats, specifies the 

Upper right corner coordinates of the door). 

OpenDoor D O 

Open or Close Door D (Integer +ve) to the Value of Door 

Closure O (Float between 0.00-1.00) where 0.00 is closed and 

1.00 is open. 

ResetMap 

Clears the contents of the MultiScape Simulation Map. This 

command must be issued prior to changing the size of room 

segments in the map. 

NoteOn M N V 

MIDI NoteOn message on MIDI Channel M (Integer 1-16), 

Note number N (Integer 1-128), Velocity V (Integer, 0-127). 

(To turn off a MIDI note send a message on the same note 

number with velocity 0). 

ProgChange M P 

MIDI Program change to program number P (Integer, 1-128) on 

MIDI channel M (Integer, 1-16). 

CtrlChange M C V 

MIDI Controller change on MIDI channel M (Integer, 1-16), to 

Controller C (Integer, 1-128), with the controller Value V 

(Integer, 0-127). 

ClvSetProgram N 

Switch Convolver program number to N (Integer, 0 to number 

of programs -1). 


TCP/IP Commands 


Connect IPAddress 

Establishes IP socket connection to machine with IP address 

“IPAddress” (String). This must be in dot notation form. 

Disconnect 

Disconnects IP socket connection previously established by the 

Connect command. 


Introduction to the Engineering 

Tools 

The Engineering Tools are a suite of Lake Technology software 

tools which enable the measurement, manipulation and realtime 

implementation of acoustic models by using the impulse 

responses of real or simulated systems. 

The Engineering Tools are comprised of two core applications: 

• The Convolver Tool, which utilises Lake Technology’s 

proprietary, low-latency convolution algorithms to enable 

the convolution of signals and impulse responses in realtime, 

using filters of up to 278,244 taps. 

— and — 

• The Measurement Tool, which is a general purpose 

impulse response measurement tool. This tool provides the 

means for, capturing and creating impulse responses from 

real acoustic spaces. The Measurement tool may also be 

used for characterising the response of audio components 

such as speakers or microphones. 

In addition to these core tools, Lake Technology provides a set 

of utilities called the SIM Tools which allow the manipulation 

of impulse responses in the *.SIM file format. 


Introduction 

Features 

Overview of the Convolver Setup 

The Convolver Tool 

The Convolver tool provides a flexible and powerful interface 

to long, low latency convolution technology. 

Each instance of the Lake Convolver requires the creation of a 

setup file, which defines a list of Filter programs. A Filter 

program is itself a definition of both the structure of a 

mathematical filter and its coefficients that are loaded from an 

impulse response file. 

The Convolver tool is organised so that users can quickly 

switch between the coefficients in the list of Filter programs 

using MIDI, keyboard or mouse inputs. This ‘fast switching’ is 

made possible by loading the Filter program coefficients into 

the DSP card’s onboard Dynamic Random Access Memory 

(DRAM). 

• The Convolver utilises low latency convolution algorithms 

to provide real-time convolution. 

• Provides up to 278,244 tap filters. 

• Fully integrated with Lake Technology’s VRack 

environment. 

• Allows fast switching between Filter programs to allow the 

A/B comparison of impulse responses. 

A Convolver Setup can be thought of as a series of nested 

elements. These nested elements are described in the diagram 

below. 


Figure 52 — Overview of the Convolver setup 

In general it can be stated that: 

THE ENGINEERING TOOLS 

• Each Convolver Setup, specified by the *.CLV file has a 

single filter type. 

• Each Convolver Setup may have from 1 to 512 Filter 

programs. 

• Each Filter program may have up 64 Filter Files (impulse 

responses, *.SIM). 


Operation 

Starting and Saving 

↓ To access the Convolver 

Figure 53 — The Convolver interface 


The Convolver tool must be loaded into the VRack from the 

New/Engineering tools submenu under New in the VRack (for 

loading instructions refer to the System Tools section of this 

manual). 

• Click the Convolver in the rack panel. 

A setup file dialogue box will appear. 

Button Function 

New Creates an untitled Convolver setup file (*.CLV). 

Open … Opens a previously saved Convolver setup file 

(*.CLV). 

Save Saves the current Convolver setup file (*.CLV), If the 

document is untitled the software will prompt with a 

Save As dialogue box. 

Save As … Save the Convolver setup file (*.CLV) under a new 

name, the software will prompt with a Save As … 

dialogue box. 

Selecting the Filter 

Once a new Setup file has been invoked, the Convolver window 

expands to offer a choice of filters. 


↓ To select the type of filter to be 

used by the Convolver 

Filter 

Type 

Number 

Filter 

Lengths in 

Taps 

Latency in 

Milliseconds 


Each filter type provides a method of accessing and utilising the 

DSP resources of the Huron system. Filters play off latency, 

length and the number of buffers available within the Convolver 

tool with the DSP resources in the system. 

1. Click the located to the right of the Filter Type Box. 

2. Select the appropriate filter from the drop down menu. 

Latency in 

Samples 

Number of DSP 

resources used 

per channel 

Number of 

Buffers available 

0 2058 0.02ms 1 1 3 

1 34728 0.06ms 3 2 3 

2 139010 0.10ms 5 3 3 

3 278244 0.10ms 5 3 1 

4 4319 1.00ms 46 1 3 

5 69422 1.00ms 48 2 3 

6 8638 2.00ms 92 1 2 

7 138924 2.00ms 94 2 2 

8 277988 4.00ms 187 2 1 

9 16384 20.00ms 968 1 15 

10 32768 41.00ms 1967 1 15 

11 65536 82.00ms 3937 1 7 

12 131072 166.00ms 7948 1 3 

13 262144 336.00ms 16110 1 1 

Tip: A single Convolver can only use 

one type of filter. To have a system 

using different types of filters, simply 

invoke more instances of the 

Convolver from the VRack. 

Creating Programs 

↓ To create a program 

Table 1 — Filters available to the Convolver 

The list contains the number assigned by Lake to the filter type, 

the length of the filter in samples, its latency in samples (ie. 

how many samples delay there is between data arriving at the 

filter and leaving the filter), its latency expressed in 

milliseconds (assuming a 48kHz sample rate), the number of 

DSP processors this filter uses when running and the number of 

buffers, or concurrent programs which can be held in memory 

to allow fast switching. 

The Convolver calls a collection of filter structures a Program. 

Up to 512 of these Programs can exist within the Convolver, 

however only one will be active at a time. 

1. Click on Add in the Program List. 


↓ To add a FIR filter to the Filter 

program 

Note: The Convolver will only accept 

SIM files which are stored as 24 bit 

binary data. 


A dialogue box appears (see Figure 54) in which the name of 

the program can be entered. 

Figure 54 — The program name dialogue box 

2. Type the name of the new program. 

3. Click OK. 

The program name will be displayed in the Program List box. 

Once a program has been created the Convolver will expand to 

show the Filter program list. The Filter program list 

represents the Finite Impulse Response (FIR) filters associated 

with each Program. Additionally, the Filter program list allows 

the outputs of a program to be mixed independently of any 

other program. 

1. Click Add in the Filter program. 

The FIR filter selection window will appear. Each FIR is stored 

in the Simulation (*.SIM) file format. 

2. Select or type the name of a FIR filter / SIM file. 

Information about the selected file is presented beneath the 

selection list box (see Figure 55). 

If a Plot has been specified in the header of the Sim file, it can 

be accessed by Clicking on Plot. 

3. Click OK. 

Figure 55 — FIR / SIM filter selection window 


↓ To remove a FIR filter from the 

filter Program. 

↓ To replace a loaded FIR filter 

Note: Each FIR filter in a Filter 

program utilises the number of DSP 

processors shown in the Filter Type 

box. The number of FIR filters which 

may be added to a program is limited 

by the number of DSP processors 

installed in the system. If the number 

of DSP processors required by the FIR 

filters exceeds those available, the 

system will return an error message 

and the excess filters will not be 

loaded. 

Inputs, Outputs, Mixing and Patching 

↓ To mix the FIR filter outputs of 

a Filter program 


The filter selection window will disappear and the selected 

impulse response will appear as part of the Filter program (see 

Figure 56 below). 

Figure 56 — Filter program window with FIR filter added 

1. Select the FIR filter. 

2. On the Filter program window Click Delete. 

1. Select the FIR filter to replace. 

2. On the Filter program window Click Edit. 

3. From the dialogue Select the replacement filter. 

4. Click OK. 

Each FIR filter within a Filter program has one input channel 

and one output channel. It is possible to mix any number of 

outputs from these filters within the Convolver. Each time a 

FIR filter is added to a Filter program, the appropriate input and 

output patch points appear on the Convolver panel in the Patch 

Bay. When the outputs of the FIR filters are mixed they are 

combined to a single output point in the Patch Bay. 

1. Select the FIR filter to be mixed from the Filter program 

window. 

2. Click Channel. 

The following window will be displayed 


Currently selected 

patch point icon 

Note: It is only possible to mix 

downwards (in number) in the output 

list, thus it is possible to mix output 2 

to output 1 but not output 1 to output 

2 

Input and Output Level Metering 

↓ To display the input and output 

levels of a Filter program 

Figure 57 — Mixing Channels 

Fast Switching with Pre-loaded Filter Coefficients 

↓ To fast switch between Filter 

programs 


3. Change the current output channel by Clicking the patch 

point icon you wish to direct the output to. 

Patch points which are no longer used will disappear from the 

Patch Bay. 

1. Select a Filter program from the Filter program window. 

2. Click Meters. 

A level meter window will appear, each meter reflects the input 

and output levels of a single filter. 

The Convolver pre-loads Filter programs and their associated 

FIR filters into on board DRAM, as they are selected. When 

two or more Filter programs and their associated FIR filters fit 

into the available DRAM they can be “fast switched”. Fast 

switching allows two different Filter programs to be compared 

in close to real time. 

It is only possible to switch between Filter programs loaded into 

DRAM, the number of filters which can be loaded into DRAM 

simultaneously is denoted by the number of buffers available 

(in Table 1 — Filters available to the Convolver). A Filter 

program which has been pre-loaded is indicated by an asterisk 

(*) next to the Program Name in the Filter program list. 

1. Click the first pre-loaded Filter program name. 

2. Click the second pre-loaded Filter program name. 


Summary 


When a Filter program is not pre-loaded it can still be switched 

to. The switching will however will take longer, as the FIR 

filters must be loaded from the disk drive. 

• The Convolver provides functionality to load and utilise FIR 

filters (SIM files) by creating Filter programs. 

• It is possible to mix outputs from FIR filters within the 

Convolver. 

• The Convolver can be used to fast switch between Filter 

programs providing a method for comparing outputs. 


Introduction 

96kHz Convolver Hardware Requirements 

The 96kHz Convolver Tool 

The 96kHz Convolver software is designed for high sample 

rate, long, low-latency convolution of 96kHz digital signals. 

The software interface to the 96kHz Convolver operates in the 

same manner as the standard Lake Convolver application, with 

the following differences: 

• To run the 96kHz Convolver, specific hardware internal and 

external to the Huron is required. 

• The 96kHz Convolver utilises filters specific to 96kHz 

convolution. 

• By virtue of the way 96kHz signals are handled in the 

Engineering tools, the appearance and manner of patching 

the 96kHz Convolver in the Patch Bay is different to that of 

the normal Engineering Tools. 

Rather than recast the standard Convolver documentation in 

terms of the 96kHz version, this sub-section focuses only on the 

differences between the two, allowing the user to substitute the 

information contained here for the relevant sections in standard 

Convolver documentation. 

The 96 kHz Convolver tool provides a flexible and powerful 

interface to high sample rate, long, low latency convolution 

technology. This high sample rate Convolver mirrors the 

functionality of the standard Lake (48kHz) Convolver, however 

it works with a 96kHz digital input signal. 

In order to run the 96kHz Convolver the following Hardware is 

required: 

• 1 Huron with at least 4 DSPs available. 

• 1 Two channel AES/EBU Digital Input Module without 

sample rate conversion (synchronous). 

• 1 Two channel AES/EBU Digital Output Module. 

• 1 96kHz Input device, which must support data formatted as 

odd and even samples in the left and right channels of a 

standard AES/EBU stream. This device may be a 96kHz 

Analogue to Digital converter. 


96kHz Filters 

Filter 

Type 

Number 

Filter 

Length 

Latency in 

Milliseconds 


• 1 96kHz Output device, which must support data formatted 

as odd and even samples in the left and right channels of a 

standard AES/EBU stream. This device may be a 96kHz 

Digital to Analogue converter. 

The 96kHz Convolver provides a selection of Finite Impulse 

Response (FIR) filters into which measured or simulated 

impulse responses may be loaded. Incoming signals are 

processed by an external 96kHz analogue to digital converter 

and are input to the Huron system via a synchronous Lake 

Digital Input Module as a set of odd and even 48kHz data 

streams. These signals are then convolved with the impulse 

responses before being output as a pair of odd and even digital 

output signals through a Lake Digital Output Module. These 

interleaved signals may then be mixed by an external 96kHz 

digital to analogue converter to produce a single convolved 

96kHz signal. 

Figure 58 — A schematic of connections to the 96kHz 

Convolver 

Rather than the standard Convolver filters the 96kHz Convolver 

uses specifically designed 96kHz filters. These filters are 

presented in the table below, and when using the 96kHz 

Convolver replace the standard Convolver filters found in Table 

1. 

Latency 

in 

Samples 

Number of DSP 

resources used per 

channel 

Number of 

Buffers 

available 

0 4116 0.02ms 2 4 3 

1 69456 0.06ms 6 8 3 

2 278020 0.10ms 10 12 3 

3 8638 1.00ms 92 4 3 

4 138844 1.00ms 96 8 3 

5 17276 2.00ms 184 4 2 

6 277848 2.00ms 188 8 2 

7 32768 20.00ms 1936 4 15 

8 65536 41.00ms 3934 4 15 

9 131072 92.00ms 7874 4 7 

10 262144 166.00ms 15896 4 3 

Table 2 — Filters available to the 96kHz Convolver 


Patching the 96kHz Convolver 

Loading the 96kHz Convolver 


For each channel of 96kHz convolution, the 96kHz Convolver 

requires an input signal which has been divided into two 48kHz 

streams of data. Each stream carries either the odd samples or 

the even samples of the 96kHz data. These odd and even data 

streams are input to the Convolver via a synchronous AES/EBU 

Digital Input Module. The 96kHz Convolver processes the 

signals in parallel, with the resultant signals being passed out of 

the Huron as a pair of odd and even 48kHz signals. The final 

96kHz output signal is derived by combining these interleaved 

samples via a Digital to Analogue converter external to the 

Huron. 

Figure 59 — Patching a channel of the 96kHz convolver correctly 

When loading the 96kHz Convolver the following guidelines 

should be used to prevent errors occurring in the output signal: 

• The 96kHz Convolver should be the first DSP using 

application loaded into the VRack. 

• The 96kHz Convolver should load all of the DSP resources 

it requires before any other applications which utilise DSPs 

are used. 

• If it is impossible for the 96kHz Convolver to be the first 

application using DSPs loaded into the VRack, it is 

important that the Convolver is started when the DSP usage 

count is a multiple of four, i.e. when the DSP usage count is 

0,4,8,12,16 etc. 


Introduction 

Features 

Operation 

The Measurement Tool 

The Measurement tool provides a method for obtaining and 

storing the impulse response (IR) of audio systems, including 

acoustic spaces (concert halls, theatres, etc) or audio processing 

equipment (reverbs, equalisers, microphones, etc). 

The measurement of an audio system requires a test signal to be 

sent to the system to be measured. Inputs to the measurement 

tool synchronously capture the signal fed back from the system. 

This signal is used to calculate the response of the system which 

is subsequently stored in a simulation (*.SIM) file. 

The response can then be analysed with SIM Tools and can be 

used as part of a Filter program within the Convolver to recreate 

the audio system measured. 

• Obtains the Impulse Response (IR) for a range of audio 

systems. 

• Generates SIM files for use in the Convolver. 

• Synchronously captures responses to test signals. 

Figure 60 — The Measurement interface 


Starting 

Note: If using measure to obtain 

impulse responses from objects 

internal to the Huron, the results may 

be offset by 2 samples from the correct 

response. This will only occur when a 

multiple channel (3 channels or more) 

measurement is performed on the 

object in question. Measurements of 

systems external to the Huron are not 

affected. 

↓ To access the Measurement tool 

Selecting the Test Signal 

↓ To select a test signal 

Note: All test files must be a power of 

two (2) in length. 

Generating the Output Signal 

↓ To transmit the output test 

signal 


The Measurement tool must be loaded into the VRack via the 

New Engineering tools submenu (for loading instructions see 

the System Tools section of this manual). 

• Click the Measurement icon in the rack panel. 

The measurement response window will appear (see Figure 60) 

Additionally, the Measurement tool will appear with its 

appropriate patchsets in the Patch Bay tool (Figure 61). 

Figure 61 — Measurement Patch Panel 

Before measurement can commence, a test signal needs to be 

selected. 

• Select a test signal, from the Signal File drop down combo 

box. 

— or — 

Scroll the Signal File drop down combo box to Other … 

then type or select the name of the test signal file. 

After selecting a test signal, the signal can be downloaded to the 

DSP system and transmitted out of a physical output. 

1. Click on Signal. 


Inputs 

↓ To change the number of Inputs 

Note: When calculating the impulse 

response, a separate impulse response 

file is generated for each input. 

Sum Multiplier 

↓ To select the Sum Multiplier 

DC Block 

↓ To enable DC Block 


The transmission of the Output Test Signal will be started, and 

the Signal button will dim, to alert the user the signal has been 

loaded. 

2. Use the Patch Bay (see the System Tools section of this 

manual) to patch the Output Test Signal to the input of the 

system under test. 

There may be a need to have more than one input to the 

Measurement tool. For example, when measuring the impulse 

response of a room, it may be desirable to take measurements at 

many different locations in the room simultaneously. Similarly, 

if a microphone other than mono (stereo, binaural, B-format, 

etc) is used, the Measure program will capture the impulse 

response on several channels synchronously. 

• Select the number of inputs required, from the Inputs drop 

down combo box. 

Once the number of Inputs is selected the Patch Bay will update 

the number of input patch points. 

The Measurement tool repeatedly outputs the measurement 

signal. Synchronously, the input samples are read at the input 

into a buffer. The impulse response is calculated from this 

buffer. The Sum Multiplier field can be used to read a number 

of sample blocks into the buffer and to average the data, which 

can give a more accurate measurement of the impulse response 

of a system in the presence of noise. 

• Select a value from the Sum Multiplier drop down combo 

box. 

DC Block can be enabled to subtract any DC offset from the 

final measurement of the impulse response. 

• Check DC Block. 


Selecting the Output File 

↓ To select the name of the 

Output File (excluding the file 

extension) 

Running the Measurement 

↓ To run the measurement 

Note: the measurement can take 

several minutes to complete, the 

bigger the Sum Multiplier and the 

more inputs there are to be measured, 

the longer the computation takes. 

Level Metering 

↓ To display level meters for all of 

the inputs and outputs to the 

Measurement tool 


The measurement of the IR of the system is written to a SIM 

file. Each input response is saved to a file specified by the 

Output File Field. The Response from the first input is written 

to the file with a .001 extension, the second input is saved to a 

file with a .002 extension, subsequent inputs follow this 

sequential naming order. 

1. Click Browse... on the Output File Field. 

2. From the dialogue box, type or select the name of the file to 

output to. 

3. Click OK. 

• Click Run. 

The Run button will become disabled until the output signal has 

been generated and an output file has been selected. 

As each input’s response is calculated an estimate of the noise 

floor is computed and displayed, this noise floor is used as a 

measurement of confidence. 

• Click Meters. 

A level meter window will appear, each meter reflects the input 

and output levels of a single signal. 


Generating Test Signals 

↓ To create the *.INV file 

Notes on Test Signals 

It is possible to generate test signals to be 

used by the Measurement Tool. Each test 

signal file saved in the SIM (*.SIM) format 

requires a corresponding inverted SIM file 

(*.INV). 

Given a test signal in a *.SIM file, the 

appropriate *.INV file can be generated by 

the Matlab function makeinv.m in the 

examples/huronmat subdirectory of the 

Huron CDROM. 


SIM File 

FFT 

Invert 

Elements 

INV File 

Figure 62 — Creating an *.INV file for deconvolution 

Several test signals are supplied with the Huron software. 

These are white and pink noises, and a special variety of swept 

sine waves. The test signals that appear automatically in the 

Measure program's list are located in the measure subdirectory. 

Any test signals (and their *.INV files) that are added to this 

subdirectory will also automatically appear in the drop down 

combo box. 

The type of test signal used will depend on the particular 

response being measured. The test signal should be at least as 

long as the duration of the system impulse response to avoid 

time domain aliasing; a good rule of thumb is to select a test 

signal twice as long as the estimated response duration. This 

should be checked after the measurement. The Measurement 

confidence indication given by the program will also help 

decide if the test signal is appropriate. 

White noise test signals (wht_NNNk.sim) should be used with 

caution when measuring electroacoustic systems containing 

tweeter speaker drivers. Since these signals contain a large 

amount of energy in the high frequencies. This energy can 

destroy tweeters with a low continuous power rating if caution 

is not exercised. If in doubt, use a low volume or switch to a 

pink noise signal. White noise is the recommended choice for 

measuring purely electrical or purely digital signal paths. While 

these test signals sound similar to Maximum Length Sequences 

(MLS) used in some other measurement tools, they will give 

better results because their non-binary nature is less likely to 

cause overshoot and distortion in D/A converters and 

transducers. 

Pink noise test signals (pnk_NNNk.sim) have a spectrum with 

equal energy per octave. This has a 3dB/octave amplitude 


Signal Generation via MatLab Scripts 

Caution 


rolloff with increasing frequency. These test signals are safer 

for electroacoustic systems and are easier on the ear than white 

noise. 

The third variety of test signals is a type of swept sine wave. 

These signals (sNN_MMMk.sim, where NN is the number of 

octaves of the sweep, and MMM is the length of the signal in 

kilosamples) contain a sinusoid who's instantaneous frequency 

sweeps up at a constant number of octaves per second until it 

reaches the Nyquist frequency (in this case, half the sampling 

frequency). This imparts the signals with two important 

properties: 

1. Pink spectrum. These signals will not damage tweeters in 

loudspeakers if used at moderate volumes. 

2. Harmonic distortion rejection. Because the time duration 

between the instantaneous excitation frequency and its 

harmonics is constant in the test signal, any harmonic 

distortion in the system under test will manifest itself as 

narrow peaks very late in the impulse response. This 

typically corresponds with a region containing no relevant 

information, and so it can be truncated (removed entirely) 

from the measurement. To take advantage of this property 

select a test signal of much greater duration than the 

response of the system under test. 

These properties mean that the swept sine test signals are the 

best choice for electroacoustic measurements, which can have 

large harmonic distortion components. 

When using these test signals, it is necessary to choose not only 

the appropriate length, but the number of octaves to be covered. 

This will depend on the frequency range of interest, and the 

physical capabilities of the test source. To find the lowest 

instantaneous frequency in a given test signal, divide the sample 

rate by: 

2 (NN+1) 

The Matlab function swptsine.m in the huronmat 

subdirectory will allow the creation of your own variations of 

these test signals. Type “help swptsine” in Matlab for 

more details. 

CAUTION: Using a sweep of too many octaves could cause 

damage to loudspeakers. These signals contain low frequencies 

at very high levels, and can cause excessive cone displacement 

in speaker drivers. If in doubt start with a signal of fewer 

octaves and visually inspect the cone displacement. The low 

frequency limit of your tests will depend on the physical 

capabilities of your speaker. Using lower volume levels can 

alleviate this problem. 


Summary 


• The Measurement Tool is a general purpose impulse 

response measurement system. 

• The Measurement Tool provides a means for generating SIM 

files for immediate use in the Convolver. 

• Various methods for creating signal files and their inverse 

are available, signal creation is particularly easy via the use 

of Lakes pre-programmed Matlab Scripts. 


Introduction 

96kHz Measurement Tool’s Hardware Requirements 

The 96kHz Measurement Tool 

The 96kHz Measurement tool provides a method for obtaining 

and storing high sample rate impulse responses (IRs) of audio 

systems; including acoustic spaces (concert halls, theatres, etc) 

or audio processing equipment (reverbs, equalisers, 

microphones, etc). This high sample rate Measurement tool 

mirrors the functionality of the standard Lake (48kHz) 

Measurement tool, with the following differences: 

• To run the Measurement 96kHz tool, specific hardware 

internal and external to the Huron is required. 

• The Patch Bay interface to the 96kHz Measurement tool is 

different to the standard Measurement tool. For each 

channel of 96kHz data to be measured Measure uses two 

input and output points corresponding to two streams of data 

representing the odd and even samples of the 96kHz signal. 

Rather than repeat the standard measure documentation with 

references to 96kHz Measure inserted, this sub-section focuses 

on the differences cited above. 

The 96kHz Measurement tool provides the same functionality, 

and utilises the same interface as Lake Technology’s standard 

Measurement tool. Its primary differentiation is that it transmits 

and captures signals at a 96kHz sampling rate, rather than the 

standard 48kHz. 

In order to run the 96kHz Measurement tool the following 

equipment is required: 

• 1 Huron with at least 4 DSPs available. 

• 1 Two channel AES/EBU Digital Input Module with no 

sample rate conversion (synchronous). One of these 

modules is required for each impulse to be measured. 


• 1 96kHz Analogue to Digital converter. This converter must 

support data formatted as odd and even samples of the left 

and right channels of a standard AES/EBU system. 



• 1 96kHz Digital to Analogue converter. This converter must 



The measurement of an audio system at 96kHz requires a test 

signal generated by the Huron to be sent from the digital output 

to a Digital to analogue converter into the system to be 

measured. The signal is returned to the Huron via an Analogue 

to digital converter and the Digital Input Module. The response 

represents the IR of the system, and can be used to create a 

finite impulse response file (*.SIM, *.001) file that may be used 

in other Lake Applications. 

Figure 63 — A schematic of connections to the 96kHz Measurement tool 

Patching the 96kHz Measurement Tool 

For each instance of 96kHz Measurement, the 96kHz Measure 

will output a signal as two 48kHz streams of data. Each stream 

carries either the odd samples or the even samples of the 96kHz 

signal. 

After being transmitted the signal is captured via a microphone 

the signal is then piped to the Measure tool from the 96kHz 

Analogue to Digital (A/D) converter and the synchronous 

AES/EBU digital input module (I/O module). Every separate 

microphone (each of which will generate an impulse response) 

requires an A/D converter and a synchronous digital I/O 

module. 

Figure 64 — Patching a channel of the 96kHz Measurement tool correctly 


Loading the 96kHz Measurement Tool 


When loading the 96kHz Measurement tool the following 

guidelines should be used to prevent errors occurring in the 

output signal: 

• The 96kHz Measurement tool should be the first DSP using 

application loaded into the VRack. 

• The 96kHz Measurement tool should load all of the DSP 

resources it requires before any other applications which 

utilise DSPs are used. 

If it is impossible for the 96kHz Measurement tool to be the 

first application using DSPs loaded into the VRack, it is 

important that the Measurement tool is started when the DSP 

usage count is 0,4,8,12,16 etc. 


Introduction to the SIM Tools 

The SIM Tools are a suite of programs which allow the 

manipulation of SIM data files. The SIM data file is the 

common file format used by all Lake Technology hardware and 

software. The SIM tools are used to convert to and from the 

SIM file format and plot, list, generate and manipulate SIM data 

files. 

The SIM tools are comprised of the applications Simwin, 

Plotwsim, Simscale, Wavtosim, and Simtowav. These 

Programs can be found in the C:\Huron32\bin\ directory 

of your installation. 

Simwin is an application which provides tools for generating 

and manipulating SIM data files. Simwin includes functions 

for: 

• Generating data, such as random noise functions. 

• Tools for the manipulation of data and vector arithmetic. 

• Generating window functions. 

Plotwsim is a SIM file plotting utility, and provides capability 

to plot files to the screen or to a printer. Plots can be made 

against time, frequency or sample numbers with logarithmic or 

linear axes. 

Simscale is a Windows based SIM scaling utility. Its primary 

function is to comparatively scale *.SIM files. This 

comparative scaling provides benefits such as: 

• The prevention of clipping of filters by scaling down large 

amplitude filters. 

• Higher signal to noise ratios by scaling up small amplitude 

filters. 

• Reduction in the amount of apparent volume difference 

when switching between filters. 

Wavtosim and Simtowav allow conversions between the SIM 

file format and the Windows WAV format for audio data 

streams. These two applications are provided on an ‘as is’ basis 

and may not work with all WAV formats. 


; comments 

( 

(dtype [bit16 | bit20 | bit24 | fixed | double] ) 

(mode [binary | text] ) 

(vsize integer) 

(dmode [periodic | stream]) 

(struct [real | imaginary | complex]) 

[(samplerate longinteger)] 

[(plotmax string)] 

[(plotmin string)] 

) 

text or binary data ... 

The SIM File Format 

The format of the *.SIM file appears below. In general terms 

the format of a *.SIM file is characterised by a header of control 

information describing the type of data and its format, followed 

by the actual data in a vector format. If the data contained in 

the vectors is complex, the real and complex parts are 

interleaved, sample by sample. 

The control information in the header is made up of the 

following components. 

Component Description 

Data type dtype set to bit16, bit20, bit24, fixed point or double 

precision. 

File mode mode is set to text or binary. 

Vector size vsize set to length of vector. 

Data mode dmode set to periodic or stream data. 

Structure struct set to complex, real or imaginary. 

Samplerate samplerate in Hertz [optional]. 

Plotmax/min string, generated internally by some Lake tools [optional]. 

Comments ; followed by comment to end of line. 

Data Formats 

The following points should be kept in mind when creating or 

modifying a *.SIM file for use with Lake Technology software. 

• Fixed point data is written as a 2's complement whole 

integer. For example, a bit16 fixed point word would be 

in the range from -32768 to 32767. Double precision 

floating point can only be uniquely represented in binary, 

and may give truncation errors when in text mode. 



• Most Lake Technology tools accept only 24 bit binary 

mode *.SIM files. 

• The plotmin and plotmax parameters contain a short 

hand form of a graphical plot of the *.SIM file. These 

parameters are interpreted by some Lake utilities to display a 

mini plot of the *.SIM file during file selection dialogues. 

Both of these parameters are optional and do not affect the 

data component of the *.SIM file. The data values for the 

parameters are generated by Lake utilities. 


Introduction 

Operation 

Starting 

↓ To start the Simwin tool 

The Simwin Tool 

Simwin provides facilities for: 

• Converting text or binary data into the *.SIM format. 

• Listing the data vectors within the *.SIM file. 

• Editing data vectors within the *.SIM file. 

• Generating various vectors (eg. random noise, sine waves, 

etc). 

• Performing vector arithmetic on data. 

• Creating window functions. 

Figure 65 — The Simwin main interface 

The Simwin application can be started from the Windows 

Explorer. 

1. From the C:\huron32\bin directory 

2. Double click on the simwin.exe icon. 

At the top of the Simwin display is the Simwin menu. 

The menu is used to open files, and launch functions. 




File Options for saving text from the main window after a 

function is executed. 

Edit Copies text from the main window into the windows 

clipboard. 

Function Loads and launches Simwin functions 

Selecting and Executing a Function 

↓ To execute a function 

Functions are used to generate, edit, or delete *.SIM files. 

Either: 

• Select a function from the Function Menu. 

—or— 

Click the appropriate Toolbar icon. 

If the function requires input, a dialogue box will appear. Each 

function utilises different dialogue boxes for data input. These 

dialogue boxes are described in the following sections of the 

manual. 

Each function command in Simwin executes the requested 

function and displays the results in the main Simwin display 

window. The text in this window may be saved to a file (using 

the Save item on the File menu) or copied to the Windows 

clipboard (using the Copy item on the Edit menu). 


Dfilter Windows Interface 

↓ To start Dfilter 

Simwin — Dfilter Tool 

The Dfilter function provides a mechanism for converting one 

data file format to another. This conversion includes 

transferring text or binary data into the SIM format, and 

converting one SIM format to another. 

Dfilter reads and writes both data of fixed point precision (up to 

32 bits in wordlength), and double precision floating point data. 

When writing fixed point data Dfilter will truncate the least 

significant bits of the data word. The input and output of all 

fixed point data used by Dfilter is written as 2's complement 

integers. 

When reading floating point data Dfilter returns a warning 

message if an attempt is made to write +1.0 as a 2’s 

complement number. If Dfilter is forced to read data outside 

the range of 2’s complement numbers, it will saturate the data, 

clipping the data outside of the valid range. 

• Select Dfilter on the Simwin function menu. 

—or— 

Click . 

This will display a dialogue box (see Figure 66). 

Figure 66 — The Dfilter interface 


↓ To convert or create a SIM file 

Note: It is possible to select a different 

sample rate for the input and output 

files in Dfilter, however sample rate 

conversion has not been supplied with 

this release of the software. Selecting 

different sample rates changes the 

information in the .SIM file header 

but does not affect the actual data 

vectors. 


1. If the input file is a SIM file then type or select the name of 

the input file. Check , the input fields will be filled 

automatically. 

— otherwise — 

Fill in the appropriate information in the input fields (data 

type, file type, sample rate) using the drop down combo boxes. 

2. Type the Vector length. 

3. If the input data is periodic Check Periodic. 

4. Specify the number of vectors to process by typing the 

number in the Vectors text box. 

5. Select if data normalisation between the input and output 

should occur, by selecting the required data type 

normalisation option (on or off). 

6. Type or select the name of the output file. 

Follow steps 1 through 4 above to fill in the appropriate 

information concerning the output file. 


Lvec Windows Interface 

↓ To start Lvec 

↓ To navigate the *.SIM file 

Simwin — Lvec Tool 

The Lvec function reads the contents of a *.SIM file and lists its 

contents in the Simwin display window. The data is presented 

in columns of real and imaginary samples, with each row 

representing a sample number. 

1. Select Lvec on the Simwin function menu. 

—or— 

Click . 

2. Select a *.SIM file to list. 

After a SIM file is selected the following dialogue box will be 

displayed (Figure 67). 

Figure 67 — The Lvec interface 

The first page of the textual display shows the header of the 

*.SIM file. The subsequent pages display the data contained 

within the *.SIM file. 

1. Click the and 

the data one page at a time. 

2. Click the 

buttons to step through 

button to go to a particular vector. 


↓ To display the magnitude of a 

complex vector 

↓ To display the data in 

hexadecimal 

↓ To display a new file 

↓ To close the Lvec function 

window 


• Check . The magnitude will be displayed in 

the real part of the vector. 

• Check . 

1. Click . 

2. Select the new file to view. 

• Click . 

This will close the Lvec function window. 


Padvec Windows Interface 

↓ To start Padvec 

↓ To pad a SIM file: 

↓ To pad inside a sample 

Simwin — Padvec Tool 

The Padvec function will read a *.SIM file and insert any 

number of zeros at any point within the file. 

• Select Padvec on the Simwin function menu. 

— or — 

Click . 

This will display the Padvec dialogue box (see Figure 68). 

Figure 68 — The Padvec interface 

1. Type the file to pad into the Input box. 

— or — 

Click Browse… and select a file name from the filename 

dialogue box. 

2. Type a filename to output to into the Output text box. 

— or — 


dialogue box. 

3. Type the sample number at which to start padding in the 

Start box. 

4. Type the number of Zeros to add after the sample in the 

Length box. 

5. Click OK to pad the Sample. 

1. Follow the steps above, and 

2. Select . 

Zeros will be inserted into the centre of the sample at the 

position specified in the start dialogue box. 


Cutvec Windows Interface 

↓ To start Cutvec 

↓ To truncate a SIM file 

Simwin — Cutvec Tool 

The Cutvec function will read a *.SIM file and truncate it to a 

specified length. 

• Select Cutvec on the Simwin function menu. 

— or — 

Click . 

This will display the Cutvec dialogue box (see Figure 69): 

Figure 69 —The Cutvec interface 

1. Type the file name of the *.SIM file to truncate into the 

Input box. 

— or — 


dialogue box. 

2. Type a filename to output the truncated file to, in the 

Output box. 

— or — 

Click Browse… and select a file name to output to from the 

filename dialogue box. 

3. Type the new length of the *.SIM file (in samples)in the 

Length box. 

4. Click OK. 


Permute Windows Interface 

↓ To start Permute 

↓ To permute a *.SIM file 

Simwin — Permute Tool 

The permute utility allows bit reversal and transposition of data 

in *.SIM files. This can be useful for the analysis of Fast 

Fourier Transform (FFT) data. 

• Select Permute on the Simwin function menu. 

— or — 

Click . 

The following dialogue box will appear (see Figure 70). 

Figure 70 — The Permute interface 

1. Type a file name to perform the permutation on, into the 

Input box. 

— or — 

Click Browse… and select an input file name from the 

filename dialogue box. 

2. Type a filename to output the permuted file to, into the 

Output box. 

— or — 


dialogue box. 

The following options for the permutation of the SIM files can 

be specified. 


↓ To produce overlap 

↓ To transpose the rows and 

columns 

↓ To reverse the direction of the 

bit data 

↓ To reverse the sequence in a 

linear mode 

↓ To Shift the sequence through a 

circular half sequence 

↓ To scale the signal through 

multiplication 

↓ To use virtual memory when 

permuting vectors greater than 

4k 


• Type a percentage in the Input Overlap and Output 

Overlap boxes. 

• Check . 

• Select a direction for bit reversal from the bit reverse drop 

down combo box. 

• Check . 

• Check . 

• Type a floating point value into the Scale box. 

1. Check . 

2. Specify the length of the columns in the Column Length 

box. 

3. Specify the length of the rows in the Row box. 

4. Specify the number of vectors in the Vectors box. 

5. Click OK to apply the permutations to the SIM file. 


Wingen Windows Interface 

↓ To start Wingen 

Simwin — Wingen Tool 

The window function generator, supports a number of different 

window functions with various mainlobe and sidelobe trade 

offs. It can create these window functions in any of the formats 

supported by the SIM file format. It will also support two 

dimensional window functions. 

The windows supported are: 

• Uniform (rectangle). 

• Bartlett (triangle). 

• Von Hann. 

• Hamming. 

• Blackman. 

• Harris. 

• Blackman-Harris. 

Wingen generates windows functions in which the peaks of the 

function are situated at the beginning and end of the sequence. 

This means the function decays to zero in the centre of the 

vector. If it is necessary to place the peak in the centre of the 

function, use the Shift option in Wingen. 

The window functions that Wingen can produce are detailed in 

the Frequency Responses for Window Functions — SimTools 

Appendix 1 sub-section of the SIM Tools. 

• Select Windows Generator on the Simwin function menu. 

—or— 

Click . 

This will display the Wingen dialogue box (see Figure 71): 

Figure 71 — The Wingen windows generator interface 


↓ To generate a window function 

↓ To generate the window 

function with a centred peak 


1. Select a window type from the drop down menu. 

2. Select the data type of the output file from the drop down 

menu. 

3. Select the file type of the output file from the drop down 

menu. 

4. Type the number of rows required in the Rows box. 

5. Type the number of columns required in the Columns box. 

6. Select the sample rate of the output file from the drop down 

menu. 

7. Type a filename to output the window file to, into the 

Output box. 

— or — 


dialogue box. 

8. Click OK. 

1. Follow the process for window generation above then: 

2. Select . 

3. Click OK. 


SigGen Windows Interface 

↓ To start SigGen 

↓ To generate a sine wave signal 

Simwin — SigGen Tool 

The SigGen function provides an interface for the generation of: 

• Sine Wave signals. 

• Noise signals. 

• Impulse signals. 

• Piecewise linear signals. 

• Select Signal Generator on the Simwin function menu. 

—or— 

Click . 

This will display the SigGen dialogue box. As different signal 

functions are selected the dialogue box will change, allowing 

the entry of function variables specific to that signal type. The 

dialogue box shown below (Figure 72) is for the generation of 

sine waves. 

Figure 72 — The SigGen interface (sine wave generation) 

1. Select the sine wave function type from the drop down 

menu. 


menu. 


menu. 

4. Type the amplitude required in the Amplitude box. 

5. Type the decibel amplitude required in the dB Amp box. 

6. Type the length of the signal in the Length box. 

7. Type the frequency of the signal into the Frequency box. 


↓ To generate a noise signal 

↓ To generate an impulse signal 


8. Type the phase of the signal into the Phase box. 

9. Type the number of vectors required into the Vectors box. 


menu. 

11. Type a filename to output the signal to, into the Output box. 

— or — 


dialogue box. 

12. Click OK. 

1. Select the Noise function type from the drop down menu. 

This will display the following dialogue box. 

Figure 73 — The SigGen interface (noise generation) 


menu. 


menu. 




7. Type the random number seed into the Seed box. 

8. Select the distribution Model of Noise from the Noise drop 

down menu. 



menu. 

11. Type a filename to output to, into the Output box. 

— or — 


dialogue box. 

12. Click OK. 

1. Select the impulse signal function type from the drop down 

menu. This will display the following dialogue box. 


↓ To generate a piecewise linear 

signal 

Figure 74 — The SigGen interface (impulse signal generation) 



menu. 


menu. 




7. Type a series of data points into the data points box. The 

data points should be input as a series of X,Y pairs (For 

example 128:1.0,150:0.8,...). 



menu. 


— or — 


dialogue box. 

11. Click OK. 

1. Select piece wise linear generation signal function type 

from the drop down menu. This will display the following 

dialogue box. 

Figure 75 — The SigGen interface (Piece wise linear generation) 


menu. 


menu. 






7. Type a series of data points into the data points box. The 

data points should be input as a series of X,Y pairs (For 

example 128:1.0,150:0.8,...). 



menu. 


— or — 


dialogue box. 

11. Click OK. 


Vector Arithmetic Windows Interface 

↓ To perform Vector Arithmetic 

Simwin — Vector Arithmetic Tool 

The vector arithmetic function of Simwin provides a means of: 

• Adding; 

• Subtracting; 

• Dividing; 

• and Multiplying vectors. 

If the datafiles contain complex data, all arithmetic performed is 

complex. 

For scaling operations, real scalar values may be specified 

instead of a second vector. 

1. Select the Vector Arithmetic on the Simwin function menu. 

—or— 

Click . 

This will display the Vector Arithmetic dialogue box (shown in 

Figure 76). 

Figure 76 — The Addvec interface 

2. Select the operation to perform from the function drop 

down menu. 

3. Type a scalar value (for scaling operations) in the Scalar 

box — this value can be used instead of a second input file. 

4. Type the number of vectors to perform the operation on in 

the Vectors box. 

5. Type the filename of the first input file, into the Input 1 

box. 

— or — 


dialogue box. 



6. Type the filename of the second input file, into the Input 2 

box. 

— or — 


dialogue box. 

— or — 

Leave blank if a scalar value has been entered. 

7. Type a filename to output the added files to, into the Output 

box. 

— or — 


dialogue box. 

8. Click OK. 


Introduction 

Features 

Operation 

↓ To start Plotwsim 

The Plotwsim Tool 

Plotwsim is a Windows based SIM file plotting utility. It 

allows the data contained in *.SIM files to be plotted in the time 

and frequency domains. 

• Plotwsim allows various display options, including 

magnitude display, and dB display. 

• Plotwsim provides the capability to zoom into and out from 

an area of interest. 

• Plotwsim allows the plot of *.SIM file to be printed. 

The Plotwsim application can be started from the Windows 

Explorer. 

1. Double click on the simwin.exe icon. 

2. From the C:\huron32\bin directory. 

3. Double Click on the Plotwsim icon. 

This will display the Plotwsim window below 

(Figure 77). 


Figure 77 — The Plotwsim main interface 


At the top of the Plotwsim main display is the Menu. This 

menu is used to open files, copy to the clipboard from the 

screen and configure the display of the SIM file. 


File Options for quitting Plotwsim and saving, loading 

and printing SIM files. 

Edit Copy the plot of the *.SIM file to the windows 

clipboard. 

View Options for configuring the plotting of a SIM file. 

Reading SIM files 

↓ To display a new SIM file 

↓ To overlay a second SIM file on 

top of the first 

Note: A Maximum of 12 SIM files can 

be overlayed on a single plot. 

1. Select New from the File menu. 

— or — 

Click . 

2. Select a *.SIM file from the Open File dialogue box. 

The SIM file will then be plotted in the Plotwsim window. 

1. Select Add from the File menu. 

— or — 

Click 

2. Select a *.SIM file from the Open File dialogue box. 

The second SIM file will be overlayed on top of the first. 


Printing 

↓ To print the SIM file plot 

↓ To select printer options 

↓ To print a border around the 

graph 

↓ To maintain relative width and 

height of the graph when 

printing 

↓ To print the graph maximising 

its height and width on the page 

↓ To maintain the original size of 

the graph 

↓ To switch the display to 

Monochrome mode 

• Select Print on the File menu. 

— or — 

Click . 


• Select Print Options on the File menu. 

A Print parameters dialogue box will be displayed (see Figure 

78). 

Figure 78 — The Plotwsim print setup dialogue 

• Check . 

• Check . 

If this option is not selected the aspect ratio may be changed to 

maximise the size of the printed image. 

• Select . 

— or — 

Select . 

1. Select . 

2. Specify the positioning of the upper left corner of the graph 

on the paper. This location is derived from two values 

which are a percentage of the width and height of the paper. 

• Select Monochrome on the View Menu. 


↓ To set up a printer 

Note: The functions 

and 

are not 

supported in this version of the 

software. 

Copying the graph to the clipboard 

↓ To copy the graph to the 

clipboard 

Viewing the Data 

↓ To view the real part of the 

complex data. 

↓ To view the imaginary part of 

the complex data 

↓ To view the magnitude of the 

complex data 

↓ To view the phase of the 

complex data 


This setting may enhance the differentiation between separate 

graphs when *.SIM files have been overlayed. 

• Select Print Setup on the File menu. 

• Select Copy on the Edit menu. 

The graph may then be pasted into a document or drawing 

program. 

• Select Real on the View Menu. 

— or— 

Click . 

• Select Imaginary on the View menu. 

— or — 

Click . 

• Select Magnitude on the View Menu. 

— or — 

Click . 

• Select Phase on the View menu. 

— or — 

Click . 


Altering the Axes 

↓ To display samples along the x 

axis 

↓ To display time along the x axis 

↓ To alter the frequency axes 

↓ To display the X axis 

logarithmically 

↓ To display the Y axis on a 

decibel scale 

• Select Sample on the View menu. 

— or — 

Click . 

• Select Time on the View menu. 

— or — 


Click . 

The time axis is calculated using the sample rate value in the 

*.SIM file header, if this is missing a sample rate of 48kHz is 

assumed. 

Frequency can be displayed along the X axis in two ways. 

The first: 

If the data contains frequency information from DC to half of 

the sample rate 

• Select Half Frequency on the View menu. 

— or — 

Click . 

The second: 

If the data contains frequency information from DC to the full 

sample rate 

• Select Full frequency on the View menu. 

— or — 

Click . 

The frequency axis is calculated using the sample rate value in 

the SIM file header, if this is missing a sample rate of 48kHz is 

assumed. 

• Select Log f on the View menu. 

— or — 

Click . 

This option is only available when displaying frequency along 

the X axis. 

• Select dB Y on the View menu. 

— or — 

Click . 


↓ To change the range of the X 

axis 

1. Select X Axis on the View menu. 

— or — 


Click . 

The X Axis alteration dialogue box will appear. 

2. Type new range numbers into the Start and End boxes. 

The value of the number is dependent on the mode in which you 

are running. 

Mode Start and End values represent 

Sample Mode The Sample Number 

Time Mode The Time in Milliseconds 

Frequency Mode The Frequency in Hertz 

↓ To change the range of the Y 

axis 

3. If a logarithmic axis is required, Select (only 

available in frequency mode). 

4. Type a number in the Step box to set the distance between 

major tick marks. 

5. Type a number in the Ticks box to set the number of minor 

ticks between two major ticks. 

6. Click OK. 

The graph with the resized axis will be displayed. 

1. Select Y Axis on the View menu. 

The Y Axis alteration dialogue box will appear. 

2. Type new range numbers into the Start and End boxes. 

The value of the number is dependent on the mode in which you 

are running. 

Mode Start and End values represent 

Sample Mode The Sample Number 

Time Mode The Time in Milliseconds 

Frequency Mode The Frequency in Hertz 

3. If a logarithmic axis is required, Check (only 

available in frequency mode). 

4. Type a number in the Step box to set the distance between 

major tick marks. 

5. Type a number in the Ticks box to set the number of minor 

ticks between two major ticks. 

6. Click OK. 

The graph with the resized axis will be displayed. 


Summary 


• Plots *.SIM file data, with a wide range of customised 

viewing options. 

• Allows direct overlay comparison of *.SIM files. 

• Provides printing control of *.SIM file plots. 


Introduction 

Features 

Operation 

↓ To start SimScale 

The SimScale Tool 

SimScale is a Windows based *.SIM file scaler. Its primary 

function is to prepare *.SIM files for use as filters in the Lake 

Convolver tool. 

SimScale is used to: 

• Prevent clipping of filters by scaling down large amplitude 

filters. 

• Provide higher signal to noise ratios by scaling up small 

amplitude filters. 

• Reduce the amount of apparent volume difference when 

switching filters by relatively scaling filters. 

• Comparative scaling using multiple input and output files. 

• Individual scaling of *.SIM files. 

The Plotwsim application can be started from the Windows 

Explorer. 

1. From the C:\huron32\bin directory. 

2. Double Click on the SimScale Icon. 

This will display a file selection dialogue. 

3. Select the directory in which the *.SIM files to be scaled are 

kept. 

Once specified a dialogue box will appear asking if all the 

*.SIM files in the selected directory are to be scaled relative to 

one another. 

4. Click Yes if you wish all the *.SIM files in the directory to 

be scaled against one another. 

5. Click No if you wish to select the 

individual *.SIM files to 

be scaled. 

6. — then — 

7. A second dialogue box will appear. From this dialogue 

select the files to be scaled. 


Note: In this version of SimScale it is 

not possible to specify the directory 

containing the *.SIM files to be scaled 

as the save directory. This is to 

prevent the corruption of the original 

*.SIM files. 

Summary 


8. Click OK. 

A dialogue box appears which indicates the amount of scaling 

to be performed on the selected files. 

9. Click OK to scale the SIM files. 

This will display a File Save… dialogue box. 

10. Specify a directory in which to save the scaled SIM files. 

11. Click OK. 

The *.SIM files will be saved and ready for use. 

• Provides a windows based interface to scale individual 

*.SIM files. 

• Provides a windows based interface to comparatively scale a 

series of *.SIM files. 


Usage 

Wavtosim 

Wavtosim is a command line application that converts 

Windows WAV files to the Lake SIM format. Invoke it by 

issuing the command wavtosim32.exe. 

wavtosim32 [-s] [-l] [-A] [-S 

start_sample] [-E end_sample] 

-s instructs Wavtosim to interpret as a SIM file 

rather than a WAV file. This is useful when combined with the 

–S or –E options. 

-l loops between the start and end samples. 

-A turns on automatic scaling for the Convolver, which scales 

the resulting SIM file down by a factor of 4. 

-S specifies the first sample to convert. 

-E specifies the last sample to convert. 


Usage 

Simtowav 

Simtowav is a command line application that converts Lake 

SIM files to the Windows WAV format. Invoke it by issuing 

the command simtowav32.exe. 

simtowav32 [-X] [-B{16,8}] [-S{48,44}] 

 

-X specifies a scale factor for the conversion where is a 

floating point number. 

-B selects either 16 or 8 bit sample resolution (16 is the default). 

-S selects either 48kHz or 44kHz sampling rate (48kHz is the 

default). 


Introduction 

Uniform Window 

Bartlett Window 

Von Hann Window 

Frequency Responses for Window 

Functions — SimTools Appendix 1 

The frequency responses of the window functions supported by 

Wingen are described in this section. 

The uniform (rectangular) window function is that which you 

see when you do not use a window function. It's characteristic 

feature is its narrow mainlobe width and its relatively high 

sidelobes. The first sidelobe is attenuated about 13dB below 

the peak of the mainlobe. The uniform window function is 

defined by equation 1. 

hn ( ) = 10 . 

N N 

n = − 

,..., −101 

, , ,..., (1) 

2 

2 

The Bartlett (triangular) window is the convolution of two 

uniform windows of length equal to half the interval. This 

implies that it has a frequency response equal to the 

multiplication of the frequency response of two rectangular 

window functions. The mainlobe has been doubled in width to 

the uniform window, and the main sidelobe is attenuated at - 

26dB. The sidelobes also decay away more rapidly. The 

Bartlett window function is defined by equation 2. 


( ) 

hn 

= 10 . − 

n 

N 

2 

N N 

n = − 

,..., −101 

, , ,..., 

2 

2 

The Von Hann is the first in a family of cos(x) windows. This 

family of windows are expressed in terms of a cosine series. 

(2)

Hamming Window 

Blackman Window 

Harris Window 


The Von Hann function can be expressed simply with only two 

terms in the cosine series. The Von Hann window function is 

defined by equation 3. 


( ) 

hn 

N n 

⎛ ⎞ 

= 05 . + 05 . cos⎜⎟ ⎝ ⎠ 

π 

N N 

n = − 

,..., −101 

, , ,..., (3) 

2 

2 

The mainlobe is twice the width of the rectangular windows and 

its highest sidelobe is attenuated relative to the mainlobe by 

32dB. 

The Hamming window is also a member of the cos(x) series 

window family. It is an extension of the Von Hann window and 

is sometimes called the generalised Von Hann window. It is a 

raised cosine of the form in equation 4. 

( ) 

hn 

N kn 

⎛ ⎞ 

= 054 . + 046 . cos⎜⎟ ⎝ ⎠ 

π 

N N 

n = (4) 

− 

,..., −101 

, , ,..., 

2 

2 

The mainlobe of the Hamming window matches the mainlobe 

width of the Von Hann, and its main sidelobes attenuate at 

43dB relative to the mainlobe. However, the sidelobes do not 

decay as rapidly as the Von Hann window. 

A cosine series with three terms, rather than the two for Von 

Hann and Hamming windows was researched by Blackman. 

One can view the generalised cosine series over K terms as that 

in equation 5. 

hn ( ) ak 

N kn 

K ⎛ π ⎞ 

= ∑ ( )cos⎜⎟ 

⎝ ⎠ 

k= 

0 

N N 

n = − 

,..., −101 

, , ,..., (5) 

2 

2 

For the Blackman window used by Wingen the coefficients 

a(0), a(1) and a(2) are shown in 6 (Next page). 

a( 

0) = 0. 42659071 

a( 

1) = 0. 49659062 

a( 

2) = 0. 07684867 

The Harris window introduces a fourth term in the cosine series 

of 5. Coefficients for the Harris window in Wingen are: 

(6)

Blackman-Harris Window 

a( 

0) = 0. 3066923 

a( 

1) = 0. 4748398 

a( 

2) = 01924696 . 

a( 

3) = 0. 0259983 


This window has the lowest sidelobe levels supported by 

Wingen. It is characterised as the minimum four term cosine 

series Blackman-Harris window. The coefficients used are: 

a( 

0) = 0. 3635819 

a( 

1) = 0. 4891775 

a( 

2) = 01365995 . 

a( 

3) = 0. 0106411 


(7)

Introduction to the Programming 

Tools 

The Programming Tools package consists of a set of function 

libraries that enable the development of both DSP programs and 

associated applications, using the Huron hardware 

environments. 

The Programming Tools documentation can be found in PDF 

format on the Huron VRack CDROM, and is comprised of 

several sub-sections: 

• The Lake Technology DSP Coding Tools explain how to 

develop DSP assembler programs for the Huron hardware 

environment. 

• The Lake Technology Debug Tools explain how to use 

LakeMon, a text based debugger which can be used for 

downloading, running, and debugging DSP assembler 

programs. 

• The Lake Technology Host Interface Library lists the C++ 

Class library structure’s used in development of host 

applications. These host applications control DSP assembler 

programs. 

• The Lake Technology Complex Library provides functions 

which enable the creation and manipulation of complex 

arrays. 

• The Lake Technology Engineering Tools Library provides 

access to Lake Technology’s FIRfilter and Transmit Capture 

classes. These classes allow application programs to 

perform convolution and measurement using the Huron 

hardware. 

• The Lake Technology Meters Class which provides 

functionality for creating and displaying a set of meters from 

within application programs. 

• The Lake Technology Debug Library provides classes for 

developing application specific LakeMon debug windows or 

including a debug window in your application. 

• The Lake Technology Error and Exception Class provides 

facilities for application programmers to handle error and 

exception conditions. 

• The Lake Technology MatLab® Interface, which provides 

the ability to control the functionality of the Huron 

Engineering Tools using MatLab Scripts. Full MatLab 

version 5.2 and 5.3 functionality is provided for the creation 

of audio engineering solutions. 


Features 

Applications 

Introduction to Equalisation Tools 

Lake EQ is a graphical equaliser which utilises “drag and drop” 

parametric filter widgets, combined with massive digital 

convolution in order to provide precision equalisation of sound 

signals, with adjustments down to 0.2Hz bandwidth. 

• Lake Technology’s “optimum phase”, giving the user a 

control for programming group delay. 

• The plotting of frequency responses provides visual 

feedback on the equalisation curve and its accuracy being 

implemented by the system. 

• >98dB signal to noise performance. 

Because of its accuracy Lake EQ can be used in a variety of 

situations some of these include: 

• Audio production. 

• Environment correction. 

• Forensic audio. 

• Acoustic research and development. 


Operation 

Starting Lake EQ 

↓ To access the EQ graphical 

interface. 

Description of the EQ Interface 

Menu bar 

Figure 79 — The Lake EQ interface 

THE EQUALISATION TOOLS 

Lake EQ must be loaded into the VRack from the 

New/Engineering Tools submenu (for loading refer to the 

System Tools section of this manual). Once loaded the EQ 

panel will appear in the VRack and an EQ patchset will appear 

in the Patch Bay. 

• Click the EQ Panel in the VRack. 

The EQ Interface will appear (see Figure 79). 

At the top of the EQ interface is the Menu Bar. The menu bar 

is used to access all the features of Lake EQ. 




File Options for saving and loading EQ canvasses, saving 

VRack workspaces, printing and exporting EQ canvasses. 

Edit Allows cutting copying and pasting of elements from the 

EQ canvass. 

View Adjusts the view settings, and allows the configuration of 

the display. 

Filters Adjusts the slider properties of an EQ canvass, and alters 

the filter response calculations. 

Graph Adjusts the display of EQ canvasses, including options for 

grid lines and logarithmic graphing. 

Setup Adjusts colours, fonts, and the limits of EQ canvasses. 

Window Standard windows organisation functions. 

Help Context sensitive help, and options. 

Dialogue Bar 

EQ Canvass 

Equalisation Curve 

Figure 80 — The EQ dialogue bar 

The Dialogue Bar contains information about the currently 

selected slider widget, group delay of the filter and the number 

of channels controlled by the canvass. If no widget is selected, 

the data boxes are blank. 

An EQ canvass is the two dimensional area upon which EQ 

slider widgets are placed. Slider widgets are graphical 

representations of the sliders used in traditional graphic 

equalisers. The bottom of the canvass is graduated in Hz 

(cycles per second) and represents the frequency range. The 

left side of the canvass is graduated in dB (decibels) and 

represents the cut/boost range. The EQ canvass may be split to 

show a view of the phase and group delay of an EQ curve. 

Such curves are used when an optimum phase EQ filter is being 

used. 

The Equalisation Curve is a graphic representation of the 

desired frequency response of the equaliser. This line is 


Slider Widget 

Toolbar 

Creating an EQ Canvass 

↓ To create a new EQ canvass 

Note: If a special EQ licence has been 

purchased and installed, it is possible 

to create up to sixteen (16) EQ 

canvasses. 


coloured in black. In cases where the hardware cannot match 

the desired equalisation response, a blue line is drawn to mark 

out the physically achieved frequency response. 

Each canvass must have one or more slider widgets in order for 

it to do any signal processing. A slider widget consists of a 

black square glued onto a vertical line and is used to modify the 

equalisation curve. 

Figure 81 — Lake EQ toolbar 

The Toolbar is used as an alternative to using the menu bar. It 

consists of a series of clickable buttons which can be used as 

shortcuts for making menu selections. 

• Select File New Canvass … 

— or — 

Click New Canvass on the tool bar (Figure 81) 

Up to four (4) EQ canvasses may be created inside the EQ 

application at one time. Each canvass may contain a different 

filter, and a different EQ response, and will consume DSP 

resources accordingly. 


Loading EQ Canvasses 

↓ To load a previously saved EQ 

canvass 

Clicking and Dropping Slider Widgets 

Band Pass 

Low Shelf 


• Select File and Open. 

Supplied with the software are two example canvasses, 

‘10band.eq’ and ‘30band.eq’. These two presets allow you to 

immediately use the Lake EQ. They are installed in the 

Huron32\EQ directory. 

Equalisation is performed by placing one or more slider widgets 

on the canvass in such a way that the desired equalisation effect 

is achieved. The vertical and horizontal placement of each 

widget on the canvass is important as their positions indicate the 

dB (vertical displacement) and frequency in Hertz (Horizontal 

displacement). 

There are three different types of slider widgets: 

A bandpass slider has a gain of 0db for all frequencies outside 

the bandwidth of the slider. Within the bandwidth of the slider 

the frequency response is a raised cosine. In short the band pass 

slider can be used as a traditional parametric equalizer. 

Figure 82 – Band Pass Equalisation 

This band pass slider is cutting the frequency response from 

about 600Hz to 1700 Hz, with a maximum cut of -15dB at 

about 1000Hz. 

A low shelf slider has a maximum gain for all frequencies less 

than the lower sideband frequency of the slider. Within the 

bandwidth of the slider the frequency response is a raised sine. 

The gain is 0db for all frequencies above the upper sideband 

frequency. 


High Shelf 

Tip: All the sliders can also be used to 

boost the signal over the frequency in 

which they operate. 

↓ To add a Slider to the canvass 

Figure 83 – Low Shelf Equalisation 


This low shelf slider starts cutting the frequency response at 

about 250Hz, hitting a low shelf point of about -14 dB at about 

120Hz. From this point the maximum cut is applied all the way 

to the lowest frequency. 

A high shelf slider has a maximum gain for all frequencies 

greater than the upper sideband frequency of the slider. Within 

the bandwidth of the slider the frequency response is a raised 

sine. The gain is 0db for all frequencies less than the lower 

sideband frequency. 

Figure 84 – High Shelf Equalisation 

This high shelf slider is performing the reverse of the low pass 

slider. The frequency response starts being cut from about 

1100Hz and drops to a maximum of about -11dB at about 

3000Hz. From this point the maximum cut is applied all the 

way to the highest frequency. 

1. Selecting the Slider widget from the toolbar. 

— or — 

Selecting a new slider from the filters menu on the menu 

bar. 


↓ To move a Slider 

↓ To adjust a Sliders bandwidth 

↓ To change the parametric 

values for a slider 

↓ To remove a Slider from a 

canvass 


2. Clicking on the position in the canvass at which the slider 

will be applied. A slider highlighted in red will appear at the 

point clicked. 

• Click and drag the Slider to its new location. 

This moves the centre frequency at which the slider operates 

and the cut/boost value to apply. 

1. Double click on a Slider. 

A red rectangle will appear around the selected Slider. 

2. Click and Drag on the rectangle handles to Increase or 

decrease the bandwidth over which the Slider will operate. 

Figure 85 — An illustration of EQ bandwidth adjustment 

The absolute value of frequency, cut/boost and bandwidth for 

the active slider are displayed in the dialogue bar: 

Figure 86 – Dialogue Bar parameters 

1. Click on the Slider to be changed. 

2. Type the values required into the dialogue bar. 

The slider will move to its new position. 

1. Click on the Slider to be deleted. 

2. Then Either: 

3. Select Delete from the Edit menu in the menu bar. 

— or — 

Press the delete key. 


Calculating the Equalisation Curve 

↓ To toggle the automatic 

calculation of responses 

↓ To manually calculate and 

download the equalisation curve 

Bypass the Equalisation Curve 

↓ To bypass the equalisation 

curve 

Setting the EQ Filter Length 


After an adjustment has been made to sliders on the EQ canvass 

the response is automatically calculated. If calculations take a 

long time, this auto-calculate feature may be turned off. With 

this feature turned off the effects of any adjustments are not 

calculated and downloaded into the hardware until the calculate 

command is given. 

Select Auto Calculate from the Filter menu. 

• Click . 

— or — 

Select Calculate Now from the Filter menu. 

— or — 

Press F9. 

The curve will be calculated and downloaded. 

In order to compare untreated signals with equalised signals it is 

possible to switch sound so that it bypasses the equalisation 

curve. 

• Click . 

This will load a flat filter in to the system, and therefore bypass 

any slider configuration for that canvass. 

Lake EQ operates using convolution. This involves the 

construction of what is known as a filter from the positions of 

all the sliders and downloading this filter to DSP boards. The 

filter represents the equalisation curve. The size of the filter 

passed to the DSP boards determines how accurately the 

equalisation curve can be modelled by the hardware. The larger 

the filter (ie. the more samples there are), the more accurate the 

resolution of the filtering that the system performs, or the finer 

that EQ adjustments can be made. 

By selecting the largest filter size, Lake EQ will always be as 

accurate as possible. Unfortunately there is a trade off: Time. 

The larger the filter, the longer it takes to construct and the 

longer it takes to download the filter to the DSP cards. 


↓ To select a filter length 

1. Select Setup filters from the Filters menu. 


Figure 87 — Lake EQ filter setup dialogue 

Low Latency Single DSP 

Filter 

Length 

Latency 

samples 

Latency 

msec. 


2. Select either a Low Latency filter or a Minimum DSP 

filter. Once selected the filter properties will be displayed. 

3. Select the filter length from the drop down Filter Length 

menu. 

4. Click OK to accept the filter settings. 

The following filters are available for use in EQ: 

DSP used Filter 

Length 

Latency 

samples 

Latency 

msec. 

2058 1 0.02 1 2058 1 0.02 1 

4319 46 1.00 1 4319 46 1.00 1 

8368 92 2.00 1 8638 92 2.00 1 

34728 3 0.06 2 16384 968 20.00 1 

69422 48 1.00 2 32768 1967 41.00 1 

138924 94 2.00 2 65536 3937 82.00 1 

139010 5 0.10 3 131072 7948 166.00 1 

278244 5 0.10 3 262144 16110 336.00 1 

Note: EQ will automatically convert 

the filters to the correct sample rate. 

The system clock can be derived 

internally, or from an external Digital 

Input Source. EQ will detect the 

system frequency and display it in the 

Sample Rate Parameter of the Setup 

- Limits menu item. 

Locking Sliders 

DSP used 

It is possible to prevent sliders from being adjusted in a 

particular way. An example of this is where a traditional 

graphic equaliser is constructed using Lake EQ. In this case the 

centre frequency or bandwidth is unable to be moved. 


↓ To lock a Slider 

↓ To unlock a Slider 

Modifying Overall Cut/Boost 

↓ To alter the overall cut/boost of 

a filter 

Zooming In and Out 


1. Select the Slider to be locked. 

2. Click the Lock check box in the dialogue bar, next to the 

option to be locked. 

— or — 

Select the appropriate option to lock from the Filter menu. 

In the example illustrated below the frequency and the 

bandwidth are locked. 

Figure 88 – Locked frequency and bandwidth 

1. Select the Slider to be unlocked. 

2. Click the Checked Lock check box in the Dialogue bar, 

next to the option to be unlocked. 

— or — 

Select the appropriate option to unlock from the Filter 

menu. 

A special slider has been included on each EQ canvass which 

allows the adjustment of the cut/boost over all frequencies. 

• Drag the special slider found at the right hand side of the 

canvass. 

Figure 89 – Cutting or boosting a filter 

Cut/Boost 

slider 

A zoom feature for canvasses has been provided to allow for 

more accurate adjustments of sliders on the canvass. 


↓ To zoom into a section of 

canvass 

↓ To zoom out of the canvass 

Saving 

Note: When saving canvasses and the 

EQ interface some of the values of the 

parametric sliders may be rounded. 

↓ To save an individual canvass 

↓ To load an individual canvass 

↓ To Save the setup of the EQ 

interface 

1. Click on the toolbar. 

— or — 

Select Zoom In from the Graph menu. 


The mouse pointer will change to when moved over the 

canvass. 

2. Select the area to enlarge by clicking and dragging the view 

rectangle across the area of the canvass to enlarge. 

3. Release the mouse button to zoom. 

• Click on the toolbar. 

— or — 

Select Zoom Out from the Graph menu. 

— or — 

Select Zoom All to return to the complete canvass view. 

Clicking this button will zoom the view out to the previous 

zoom position. 

There are two methods of saving EQ setups. The first, using 

*.EQ files saves individual canvasses, while the second using 

*.WSQ files saves all the canvasses and the setup of the EQ 

interface. 

1. Select the canvass to be saved. 

2. Select Save As from the File menu on the Menu Bar. 

— or — 

Click the save icon on the toolbar (see Figure 81). 

1. Select Open from the main menu. 

2. Select the file to load. 

3. Click OK. 

The canvass will be loaded into the EQ interface. 

Alternatively it is possible to save the entire EQ workspace, 

including preferences, window size, toolbar positions and all 

currently loaded canvasses. 

1. Select Save Workspace from the File menu on the menu 

bar. 

2. Enter a name to save the workspace file as. 


Limits 

↓ To alter the graph limits 

↓ To apply the changes to chart 

limits to all of the canvasses in 

the EQ 

↓ To reset the chart limits to the 

default values 

Phase Considerations 


3. Click OK. 

The workspace file will be saved with all settings intact. If any 

canvasses are unsaved the EQ program will prompt to save each 

one. 

It is possible to configure Lake EQ’s graph limits from the 

menu bar. 

1. Select Limits … from Setup on the menu bar. 


Figure 90 — Lake EQ limits dialogue 

2. Change the limits by typing the required values into the 

dialogue boxes. 

• Check Apply to all. 

1. Click Default. 


Lake EQ can run convolution without propagation delay. In 

order to achieve this the EQ imposes group delay in phase 

linear filtering. In order to reduce or remove this group delay 

Lake has introduced the “optimum phase”. Using the optimum 

phase, filters can be programmed to have zero or minimum 

phase delay, the trade-off however is an increase in phase 

distortion. Phase distortion can be reduced dramatically by 

selecting a phase delay value slightly larger than zero, for 


↓ To set Lake EQ to use Optimum 

Phase 

Exporting and Importing Equalisation Data 

↓ To export an EQ or impulse 

response file 

↓ To import an EQ or impulse 

response file 

Stopping EQ 

↓ To exit Lake EQ 

Summary 


example a delay of 2msec will improve most of the phase 

distortion. 

1. Select Optimum Phase from Design Criteria on the filter 

menu. 

2. Set the delay required in the Delay (secs) box in the 

Dialogue bar. 

It is possible to export impulse responses and EQ data as text. 

1. Select Export from the File menu 

2. Select the Impulse response or EQ data to save as text. 

3. Save the file with the extension .EQX. 

4. Click OK. 

1. Select Import from the File menu 

2. Select the Impulse response or EQ data to import. 

3. Click OK. 

• Click Close. 

— or — 

Select Exit from the File menu. 

• Lake EQ provides features for a wide range of equalisation 

tasks. It is highly user configurable and provides a range of 

accuracy in equalisation curves which meets the 

requirements of even the most stringent equalisation 

applications. 


Introduction 

96kHz EQ Hardware Requirements 

The 96kHz EQ 

Lake 96kHz EQ is a graphical equaliser which utilises “drag 

and drop” parametric filter widgets, combined with massive 

digital convolution in order to provide precision equalisation of 

sound signals, with adjustments down to 0.72Hz bandwidth. 

The software interface to the 96kHz EQ operates in the same 

manner as the standard Lake Technology EQ application, with 

the following differences: 

• To run the 96kHz EQ, specific hardware internal and 

external to the Huron is required. 

• The 96kHz EQ utilises convolution filters specific to 96kHz 

equalisation. 

• By virtue of the way 96kHz signals are handled by the 

Huron, the appearance and manner of patching the 96kHz 

EQ in the Patch Bay is different to that of the normal EQ 

tool. 

This sub-section illustrates the differences between the standard 

Lake EQ and the 96kHz version. It is left up to the user to 

substitute the information contained here for the relevant parts 

of the standard EQ documentation. 

Lake EQ 96kHz is a graphical equaliser which utilises “drag 

and drop” parametric filter widgets, combined with massive 

digital convolution in order to provide precision equalisation of 

sound signals, with adjustments down to 0.72Hz bandwidth. 

In order to run the 96kHz EQ the following Hardware is 

required: 

• 1 Huron with at least 4 DSP’s available. 

• 1 Two channel AES/EBU Digital Input Module with no 

sample rate conversion (synchronous). 


• 1 96kHz Analogue to Digital converter. This converter must 




96kHz Filters 


• 1 96kHz Digital to Analogue converter. This converter must 



The 96kHz EQ utilises customised 96kHz convolution filters to 

undertake its signal processing. 

Incoming signals are processed by an external 96kHz analogue 

to digital converter and are input to the Huron system via a 

synchronous Lake Digital Input Module as a set of odd and 

even 48kHz data streams. These signals are then convolved by 

EQ 96kHz with the responses in the EQ canvasses before being 

output as a pair of odd and even digital output signals through a 

Lake Digital Output Module. These interleaved signals may 

then be mixed by an external 96kHz digital to analogue 

converter to produce a single Equalised signal. 

Figure 91 — A schematic of connections to the 96kHz EQ 

Low Latency Minimum DSP 

Length Latency 

Smpl. 

Latency 

Secs. 

The 96kHz EQ utilises specific 96kHz filters during 

convolution. These filters have the following properties, and 

should be used in place of the standard EQ filters when running 

the EQ 96kHz. 

The following filters are available to the 96kHz EQ: 

DSP Length Latency 

Smpl. 

Latency 

Secs. 

4116 2 0.02 4 4116 2 0.02 4 

8638 92 1.00 4 8638 92 1.00 4 

69456 6 0.06 8 17276 184 2.00 4 

138844 96 1.00 8 32768 1936 20.00 4 

278020 10 0.10 12 65536 3934 41.00 4 

131072 7874 82.00 4 

262144 15896 166.00 4 

Patching EQ 96kHz 

DSP 

For each I/O of 96kHz EQ, the EQ 96kHz requires an input 

signal that has been divided into two 48kHz streams of data. 


Loading EQ 96kHz 


Each stream carries either the odd samples or the even samples 

of the 96kHz data. These odd and even data streams are input 

to the EQ via a synchronous AES/EBU digital input module. 

The EQ 96kHz processes the signals in parallel, with the 

resultant signals being passed out of the Huron as a pair of odd 

and even 48kHz signals. The final output signal is derived by 

combining these interleaved samples via a digital to analogue 

converter external to the Huron. 

Figure 92 — Patching the EQ 96kHz tool correctly 

When loading EQ 96kHz the following guidelines should be 

used to prevent errors occurring in the output signal: 

The 96kHz EQ should be the first DSP using application loaded 

into the VRack. 

The 96kHz EQ should load all of the DSP resources it requires 

before any other applications that utilise DSP’s are used. 

If it is impossible for the 96kHz EQ to be the first application 

using DSP’s loaded into the VRack, it is important that EQ 

96kHz is started when the DSP usage count is 4,8,12,16 etc. 


Other Tools 

This section introduces a series of utility applications that Lake 

Technology provides to its customers. 

VMixer, This application is a versatile mixing and audio 

routing application that can be controlled remotely via SNAP 

commands. 

ABSwitcher, This application provides basic functionality for 

switching between multiple inputs / applications enabling users 

to hear AB comparisons without requiring the repatching of 

applications. 

BFormat Mixer, This application mixes two BFormat inputs, 

and provides a single BFormat encoded output stream. 

I/O Test and I/O Test 96k, These applications allow a user to 

test the Inputs and Outputs of the Huron system. 


Features 

Operation 

The VMixer 

The VMixer application is a versatile mixing and audio routing 

application that can be controlled remotely by SNAP (Spatial 

Network Audio Protocol) commands. 

• Resizable number of inputs and outputs with up to 24 I/O 

channels. 

• Each input can be routed to any number of individual 

outputs. 

• Signal level can be controlled individually on each input and 

output. 

• Grouping of level controls. 

• Signal level metering on every input and output. 

• Labelling of individual inputs and outputs. 

• Remote control with the Socket application (via SNAP). 

Figure 93 — The VMixer interface 


Starting 

↓ To access the VMixer 

↓ To start the default VMixer 

configuration 

Opening and Saving Settings 

↓ To save the current VMixer 

settings 

↓ To load VMixer settings 

Resizing 

↓ To set the number of input 

channels 

↓ To set the number of output 

channels 

OTHER TOOLS 

The VMixer must be loaded into the VRack through the 

New/Others Tools submenu (for loading instructions see the 

System Tools section of this manual). Once loaded, the VMixer 

will appear in the VRack, and its associated patchset will appear 

in the Patch Bay. 

• Click on the VMixer panel in the VRack. 

• Click on the Default button on the VMixer window. 

1. Click on the Save button. 

2. Browse for the directory you wish to save your settings in. 

3. Type in a file name. 

4. Click Save. 

A new file with the .VMX file extension will be created in the 

selected directory. 

1. Click Open. 

2. Browse for the settings file (with file extension .VMX) that 

you wish to open. 

3. Click Open. 

VMixer can be resized to include up to 24 input and output 

audio channels. 

1. Change the no. inputs setting to the desired number of input 

channels required. The no. inputs must be an integer value 

between 1 and 24. 

2. Click Apply. 

1. Change the no. outputs setting to the desired number of 

output channels required. The no. outputs must be an 

integer value between 1 and 24. 



Note: When increasing the size of 

VMixer, the extra channels added 

during the resize operation are 

initialised to the default channel 

level and audio routing settings. 

Labelling Channels 

↓ To label an input channel 

↓ To label an output channel 

↓ To make labels appear in the 

patchbay 

Routing Audio 

↓ To route audio from an input to 

an output 

↓ To remove an audio routing 

node 

OTHER TOOLS 

Each of VMixer’s input and output channels can be labelled for 

easy reference. 

If input level meters are currently displayed, click on the input 

label / meter toggle button to display input labels. 

1. Double click on the input label you wish to name. 

2. Type the name into the dialog box. 

3. Click OK to return to VMixer. 

If output level meters are currently displayed, click on the 

output label / meter toggle button to display output labels. 

1. Double click on the output label you wish to name. 

2. Type the name into the dialog box. 

3. Click OK to return to the main VMixer interface. 

• Click Apply. 

VMixer includes a flexible routing matrix allowing any input to 

be connected to any number of individual outputs. Each output 

is connected to an input by an audio routing node, and each 

audio routing node can also be assigned a gain. 

1. Place the mouse over the audio routing matrix. 

2. Vertically align the mouse with the input channel that you 

wish to route audio from. 

3. Horizontally align the mouse with the output channel that 

you wish to route audio to. 

4. Click to place an audio routing node. 

The default gain for an audio routing node is unity (0 dB). 

• Double click on the audio routing node. 

OR 

1. Right click on the audio routing node. 

2. Click disconnect. 


↓ To adjust the gain of an audio 

routing node 

↓ To set the gain of an audio 

routing node back to unity 

Adjusting Level 

↓ To adjust the level of an input 

channel 

↓ To adjust the level of an output 

channel 

Grouping Level Controls 

↓ To group input channel levels 

together 

↓ To group output channel levels 

together 

Note: Input and output levels 

cannot be part of the same group. 

↓ To remove grouping 

OTHER TOOLS 

1. Right click on the audio routing node. 

2. Move the slider in the Gain Control dialog box to the new 

level. Moving the slider to the left will result in increased 

gain, moving the slider to the right will result in a reduction 

in gain. 

3. Click OK to return to the main VMixer interface. 

1. Double click on the audio routing node to remove it. 

2. Single click to create a new audio routing node with unity 

gain. 

• Click on an input level slider and drag it to the desired level. 

Maximum signal level is obtained when the slider is moved 

to the top of its travel. 

• Click on the output level slider and drag it to the desired 

level. Maximum signal level is obtained when the slider is 

moved to the far left. 

1. Click the input channel select button for all of the channels 

to be grouped. As each input channel select button is 

pressed it will glow. 

2. Click the Group button. 

1. Click the output channel select button for all of the 

channels to be grouped. As each output channel select 

button is pressed it will glow. 

2. Click the Group button. 

When channel levels are grouped together, a change to the level 

of one channel will affect all channels in the group. 

1. Click the input channel select or output channel select 

buttons for all of the channels to be removed from a group. 


Muting 

↓ To mute all output channels 

↓ To unmute all output channels 

Displaying Level Meters 

↓ To display input level meters 

↓ To display output level meters 

Exiting VMixer 

↓ To quit the VMixer 

DSP Usage 

2. Click the Ungroup button. 

OTHER TOOLS 

• Click on Mute. The mute button will be surrounded in red to 

indicate that mute has been enabled. 

• Click on Mute again. 

VMixer includes level meters on each audio input and output 

channel. Meter range is from –40dB to 0dB. By default, 

channel labels are shown in place of channel meters. 

• Click the input meter / label toggle button. 

Input level meters will be displayed on the screen in place of the 

input channel labels. Input level meters display the signal level 

before the gain of the output level control is applied. 

If the level meters are already displayed, clicking on the input 

meter / label toggle button again will cause the input labels to 

be displayed. 

• Click the output meter / label toggle button. 

Output level meters will be displayed on the screen in place of 

the output channel labels. Output level meters display the 

signal level after the gain of the input level control is applied. 

If the level meters are already displayed, clicking on the output 

meter / label toggle button again will cause the output labels 

to be displayed. 

Click the close control button on the menu bar of the VMixer 

window. 

The VMixer’s DSP resource usage is dependent on the number 

of input and output channels that are available, as shown in the 

table below: 


Controlling VMixer Remotely 

Summary 

Resources No. Input 

Channels 

1 DSP 1-12 1-12 

2 DSP 13-24 1-12 

No. Output 

Channels 

2 DSP 1-12 13-24 

4 DSP 13-24 13-24 

OTHER TOOLS 

VMixer can be controlled remotely with SNAP (Spatial 

Network Audio Protocol) and the Socket application, 

facilitating complex audio patching and level control to be 

initiated by an external device connected to the same TCP/IP 

network. Refer to the SNAP section for more information. 

• VMixer is an audio routing and mixing application. 

• Each input and output level can be controlled individually. 

• The gain of each audio routing node can be controlled 

individually. 

• VMixer can be resized to include up to 24 audio input and 

output channels. 

• VMixer can be controlled remotely using the Socket 

application (via SNAP). 


Operation 

Starting 

↓ To access the AB Switcher 

Selecting an Output Pair 

ABSwitcher 

The AB Switcher provides a fast switching mechanism for 

undertaking AB comparisons of headphone related applications. 

The AB Switcher uses one DSP worth of resources. 

Figure 94 — The AB Switcher interface 

The AB Switcher must be loaded into the VRack through the 


System Tools section of this manual). Once loaded, the AB 

Switcher will appear in the VRack, and its associated patchset 


• Click on the AB Switcher panel in the VRack. 

The AB Switcher window will appear (see Figure 94). 

The AB Switcher takes a series of inputs and presents them to a 

set of outputs. In order to do A to B comparisons it is possible 

to switch the output between input pairs. 


↓ To select which pair of inputs 

will be output 

Displaying Meters 

↓ To display metering of inputs 

into and out of the AB switcher 

Exiting AB Switcher 

↓ To quit the AB Switcher 

Summary 

OTHER TOOLS 

• Click the Input pair corresponding to the input pair 

required. 

The sound from these inputs will be played through the L R 

outputs of the AB Switcher. 

• Click meters. 

A window representing metering on the inputs and outputs of 

the AB switcher will appear. 

• Click the Close control button on the menu bar of the AB 

Switcher window. 

• The AB Switcher allows quick comparisons between 

headphone related inputs and outputs. 

• The AB Switcher allows metering of multiple inputs and 

outputs. 

• The AB Switcher allows multiple listeners to hear the same 

outputs through identical multiple outputs. 


Operation 

Starting 

↓ To access the BFormat Mixer 

Altering Gain Levels 

↓ To alter the gain levels of the 

inputs 

BFormat Mixer 

The BFormat Mixer mixes two BFormat Signals separated into 

the X,Y,Z,ω format and mixes them to a single set of BFormat 

X,Y,Z,ω outputs. 

Figure 95 — The BFormat mixer interface 

The BFormat Mixer must be loaded into the VRack through the 


System Tools section of this manual). Once loaded, the 

BFormat Mixer will appear in the VRack, and its associated 


• Click on the BFormat Mixer panel in the VRack. 

The BFormat Mixer window will appear (see Figure 95). 

1. Select either the input 1 or input 2 gain level box. 

2. Type the new gain value in dB (this number must be 

negative). 



Altering the XYZ Balance 

↓ To alter the XYZ Balance levels 

of the inputs 

Altering the Azimuth 

Azimuth can be defined as the facing 

to an object on a horizontal plane. 

Values start at 0.00 for forward and 

increase as the object is further to the 

left, up to 360.00 

↓ To alter the Azimuth of the 

inputs 

Altering the Elevation 

↓ To alter the Elevation of the 

inputs 

Viewing BFormat Mixer Metering 

↓ To view the BFormat Mixers 

metering 

OTHER TOOLS 

The XYZ balance in the BFormat mixer represents the gain for 

the XYZ portions of the BFormat signals. 

1. Select either the input 1 or input 2 XYZ Balance box. 

2. Type the new XYZ balance value in dB (this number must 

be negative). 


The Azimuth of a BFormat signal represents the direction a 

sound appears to be coming from on a horizontal plane. 

1. Select either the input 1 or input 2 Azimuth level box. 

2. Type the new azimuth value. 


The elevation of a BFormat signal represents the apparent 

vertical angle from which the sound comes. 

1. Select either the input 1 or input 2 Elevation box. 

2. Type the new elevation value. 


The BFormat Mixer provides metering for the inputs and 

outputs of the BFormat Mixer. 

• Click Meters. 

The BFormat meters window will appear. 


Summary 

OTHER TOOLS 

• The BFormat Mixer mixes two BFormat X,Y,Z,ω inputs into 

a single X,Y,Z,ω output. 

• The Azimuth, Elevation and Gain of the inputs may be 

altered through the mixer to produce the final output data. 


Operation 

Starting 

↓ To access I/O Test 

I/O Test and I/O 96k Test 

I/O Test and I/O Test 96k are designed for testing the Input and 

Output modules of the Huron system. I/O test produces Sine 

waves and White noise and provides output to the screen and to 

a logfile for later analysis. 

Figure 96 — The IO test interface 

I/O Test and I/O Test 96k must be loaded into the VRack 

through the New/Others Tools submenu (for loading 

instructions the System Tools section of this manual). Once 

loaded, I/O Test will appear in the VRack, and its associated 

patchset will appear in the Patcher tool. To utilise I/O Test and 

I/O Test 96k properly you will require a Male/Female XLR 

cable to connect the I/O modules to be tested together. 

• Click on the I/O Test panel in the VRack. 

The I/O Test window will appear (see Figure 96). 


I/O Test Patching 

Selecting the Format of Results 

↓ To output to a log file 

↓ To output a summary only to a 

log file 

↓ To output only to the I/O test 

window 

OTHER TOOLS 

The I/O Test requires software patching via the Patch Bay as 

well as physical patching on the Huron Patch Panel for it to 

function properly. The Software and Hardware patching must 

coincide with the modules to be tested. 

The patching example given below shows how to test the Input 

15 and Output 3 modules. 

Figure 97 — The correct patching for IO Test 

Once patched the I/O Tests may be run. 

Each of the tests described below have a range of output 

formats available. These outputs may be set through the output 

selection boxes found below the signal section of the I/O test 

window. 

1. Select Log. 

2. Click File. 

3. From the Selection window enter a new file name or choose 

a file to save the log information to. 

4. Run the Test (see below). 

1. Select Summary only. 

2. Click File. 

3. From the Selection window enter a new file name or choose 

a file to save the Summary log information to. 


1. Deselect Log. 


The Test results will only appear in the display window. 


Testing Signal to Noise Ratios 

↓ To test the signal to noise ratio 

of Inputs and Outputs 

1. Select Sine. 

2. Type a Frequency for the Sine wave. 

3. Type an Amplitude for the Sine wave. 

4. Enter a sum multiplier. 

5. Click Gen Output to generate the output signal. 

6. Click RunTest to start the test and output the results. 

The results will be displayed in the following format. 

OTHER TOOLS 

Report Result 

Sine Signal Level in dB should be equivalent to input 

Sine Frequency In Hz should be equivalent to input 

Signal to Noise ratio + Distortion In dB should be approximately 80dB 

DC offset In dB should be approximately -66.00dB (however this can 

vary widely). 

Testing I/O’s for Deviation from Normal Frequency Characteristics 

↓ To test deviation in the 

frequency characteristics of 

input and output modules 

1. Select White Noise. 

2. Type an amplitude. 

3. Select a Simfile to output the white noise to if required. 

4. Click Gen Output to generate the output signal. 

5. Click RunTest to run the test. 

The results will be displayed in the following format: 

Hz dB Deviation Hz dB Deviation 

23.44 35.16 

58.59 128.91 

246.09 503.91 

996.09 2003.91 

3996.09 8003.91 

15996.09 20003.91 

The value for the deviation should be less than (.5 dB.), if it is 

significantly higher than this then one of the modules may be 

faulty. 


Summary 

• To utilise I/O Test an external patch cable is required. 

OTHER TOOLS 

• I/O Test allows testing of both signal to noise ratios, and the 

testing of expected characteristics of Input and Output 

modules. 


Huron Technical Reference 

The Huron technical reference section contains information 

concerning the technical specifications of the Huron and 

information to aid users in the correct use of signals and data 

formats with the Huron system. The following sub-sections are 

covered: 

• Data Formats and Scaling, which discusses the format of 

data used in the Huron system, and the most effective way to 

scale signal files to obtain the best performance from the 

Huron. 

• Hardware Specifications, provides a set of tables outlining 

the hardware present in the Huron. 

• Managing Audio Signal Levels, examines the effective use 

of the VRack, IO Manager and equipment external to the 

Huron, in order to obtain the optimal audio performance 

from the system. 

• The Glossary, contains a list of common words and 

acronyms found throughout this manual. 


Fixed Point Digital Audio Data Representation 

Data Formats and Scaling 

Considerations 

Digital audio data within the Huron system is transferred over 

the Huron bus in 24-bit twos-complement format. This data 

format provides for representation of audio data with Root- 

Mean-Square (RMS) quantisation at a level of 146.3dB below 

the RMS level of a full-scale sine wave. The range of 24-bit 

values is as follows: 

Hexadecimal 

value 

Decimal value Signal Level Description 

0x7FFFFF 8388607 0.99999988 Positive full scale 

→ → → → 

0x000001 1 0.00000012 Smallest positive value 

0x000000 0 0.00000000 Zero level 

0xFFFFFF -1 -0.00000012 Smallest negative value 

→ → → → 

0x800000 -8388608 -1.00000000 Negative full scale 

Finite Impulse Response Filter (FIR) Coefficient Scaling 

Filter Scaling for Unity Gain 

To gain optimum performance from the system, the following 

files must be scaled appropriately: 

• FIR filter coefficients which are loaded into the Huron 

system via *.SIM data files. The data in these files is stored 

as 24—bit fixed point data. 

• Time domain impulse response filters used by the Huron 

Convolver tool. 

If necessary, *.SIM files can be re-scaled using the SIMWIN 

utility MultVec or SimScale. 

There are two important factors to consider during the 

generation or re-scaling of data files: 

FIR filter coefficient files should ideally be scaled so that the 

audio data that is used in typical listening tests will pass through 

the FIR filter with approximately unity gain. Coefficient files 

that are scaled up too high will result in a filter which boosts the 

output level, possibly resulting in clipping at the FIR filter 

output when the input signal is close to full scale. Coefficient 


HURON TECHNICAL REFERENCE 

files that are scaled down too low will produce output from the 

FIR filter that is well below full scale, resulting in poor signalto-noise 

performance in the D/A converter. 

The diagram below illustrates the way filter coefficient scaling 

effects the Convolver’s output level: 

Figure 98 — Scaling of filter output due to filter coefficients 

The amplitude of the filter coefficients (h(k)) determine the 

overall gain of the filter. More specifically, the filter scale (S) is 

defined as: 

N−1 

1 

S = ∑ h( 

k ) 

2097152 

k= 

0 

As a reference for comparison, the Huron Convolution Tools 

are supplied with an example file called 

c:\Huron32\sim\unity.sim. This file specifies an all-pass, 

unity gain filter. It contains one impulse in the first sample of 

the vector, and all zeros following this. The magnitude of the 

impulse is 2097152, which results in a gain of S=1. 

For most real filter responses, the overall gain S of the filter is 

mostly useful in defining the average gain of the filter, for white 

noise audio input. In reality, typical audio signals have most 

energy in a narrower band (from about 200Hz to 8000Hz). For 

this reason, it can be more meaningful to compute the gain S 

based on a band-limited portion of h(k). 

In order to achieve optimum data scaling in the Huron system, it 

is advised that all SIM files be scaled up to achieve a gain of 

S=1. This can be done by the following procedure : 

• Determine the gain (S) of a SIM file (or a collection of SIM 

files that are to be used together). Determine the gain for a 

band-limited portion of the SIM file. 

• Use the MultVec utility to scale the data up or down by a 

factor of 1/S. This means that the newly scaled SIM file 

should now have an average gain of S=1. 

• If a collection of SIM files are to be used together, then the 

relative amplitude difference between their gains must be 

preserved, so the files should all be scaled up by the same 

amount. The Lake Technology SimScale tool described 

below uses the formula 1/max(Sn) to achieve this result. 


2

↓ SimScale 

Avoiding Large Frequency Peaks 


The revision 1.0 Huron Convolver Filter is not capable of 

boosting any one frequency by more than 12dB This may make 

it impossible to scale up the overall filter to unity gain. If the 

frequency response of the filter (i.e. the FFT of h(k)) contains 

large, narrow peaks that are more than 12dB above the average 

frequency response level, then it will not be possible to scale the 

filter up to unity gain (S=1). 

SimScale is a windows program supplied with the Huron 

Engineering tools. It is used to scale a filter or a group of filters 

to unity gain. The program will either scale the filters so that 

the largest filter has unity gain, or the largest peak is no more 

than 12dB above unity. 

The function and operation of SimScale is described in the 

SimTools section of this manual. 


Huron Technical Specifications 

16 Bit A/D Modules 

Input Impedance 10kOhm balanced 

Input Gain +91.5dB -95.5dB 

Maximum Input Level +20dBm @ 0dB gain, +0dBm @ 20dB gain 

Group Delay 0.375ms (18 Sample periods) 

Group Delay Linearity 0.0us 

18 Bit D/A Modules 

Output Impedance 300 Ohm balanced, floating type 

Output Gain +31.5dB ⇔ -95.5dB 

Maximum Output Level +20dBm 

Group Delay 

Phase Linearity 

Digital Input Specifications 

0.687ms (33 Sample periods) 

< 0.5degrees 

Input Impedance and Level 110 Ohm - Professional Levels 

Formats Supported 

AES/EBU, IEC 958, S/PDIF, EIAJ CP-340 

Input Clock Range 25kHz-55kHz 

Digital Input Specification with Sample Rate 

Converter 

Input Frequency Ratio 1:2 ⇒ 2:1 

Total Harmonic Distortion + Noise > 94dB 

Dynamic Range 

Digital Output Specifications 

> 120dB 

Output Impedance 110 Ohm Professional and 75 Ohm Consumer 

Output Format 

System Performance (A/D -> D/A) 

AES/EBU, IEC 958, EIAJ CP-340 

Total Harmonic Distortion < 0.0025%, 1kHz, 0dBu 

Dynamic Range > 90dB 

SNR 90dB 

Crosstalk < -90dB @ 1kHz < -80dB @ 20kHz 

Sampling Rate 48kHz 

Frequency Response, 20Hz-20kHz +0.1dB -0.5dB 

Group Delay 

System Specifications 

1.1ms + DSP Delay (53+DSP Sample Period Delay) 

Environment 0-50°C Operating 

Power Requirements 90-130 Vac 220-260 Vac 

Power Consumption Configuration Dependent, 200W Max 

Net Weight 

~23kg 

Performance Plots 

Below is a plot showing the harmonic distortion and the Signal 

to Noise Ratio (SNR) of the 16 bit A/D and 18 bit D/A Huron 

System. The plot was derived by presenting a very pure 1kHz 

sine wave to the A/D and D/A, and analysing the output with a 

4k Fast Fourier Transform (FFT). 


Figure 99 — SNR of the 16 bit A/D and 18 Bit D/A 

SNR=92.08dB 

S/(N+D)=92.02dB 


Below is a plot illustrating the dynamic range of a typical 16 bit 

A/D and 18 bit D/A Huron system. This plot was derived by 

presenting a very pure low-level sine wave to the system. 

Figure 100 — Dynamic Range of the 16 bit A/D and 18 bit D/A 

Dynamic Range = 93.08dB 

The plot below shows the frequency response of the 16 bit A/D 

and 18 bit D/A Huron system. 


Figure 101 — Frequency Response of the 16 bit A/D and 18 bit D/A 


The plot below is a diagram showing the system latency when a 

“zero latency” filter is being run. The total sample period delay 

in this system is 54 sample periods, with a one sample delay for 

the FIR. 

Figure 102 — System Latency of a zero latency filter 

The latency of the 278,244 tap filter is illustrated below. The 

latency of the actual DSP filter function is 5 samples, yielding 

an overall system delay of 58 samples. 


Mechanical Specifications 

1 

3 

Figure 103 — Latency of the 278,244 tap filter 

2 


The patch panel XLR connector specifications are illustrated 

below. 

Figure 104 — XLR connectors 

2 1 

Audio Out connector (male) Audio in connector (female) 

1. Shield. 

1. Shield. 

2. Signal positive. 2. Signal positive. 

3. Signal return. 3. Signal return. 


3

D-62 Connector Specifications 


The diagram below illustrates the I/O board D-62 connector 

specifications. 

Figure 105 — I/O Board D-62 connector specifications 


Audio Input Levels 

Managing Audio Signal Levels for 

Optimum Noise Performance 

Optimal noise performance can be achieved when using a 

Huron System by ensuring that both analogue and digital 

signals are kept as close as possible to full scale throughout 

each stage of processing. 

Analogue audio input to the system is converted to digital data 

by the A/D module. This module contains analogue circuitry 

that allows for a variety of different input types (balanced and 

unbalanced) and different signal levels (microphone, consumer 

line level and professional line level). A simplified schematic 

of the A/D input is shown below. 

Figure 106 — A/D input schematic 

The volume control settings on the A/D modules are 

manipulated by the lower row of sliders in the I/O Manager. 

The pre-amplifier settings are selected by the buttons (labelled 

0dB, 20dB, 40dB and 60dB) in the I/O Manager. 

The following recommendations apply to the use of the A/D 

modules: 

• Always supply the highest level analogue input possible to 

the A/D module (but not exceeding 20V p-p). This will 

ensure that the signal level is maintained at its maximum at 

all stages of the analogue input circuit, thus reducing the 

contribution of noise effects. 



• Always use the pre-amplifier in preference to the volume 

control to boost lower level analogue inputs. It is always 

preferable to increase the analogue signal level as much as 

possible at the pre-amplifier to avoid an unnecessary noise 

contribution at the input to the volume control. 

• Always aim to provide a near full scale input signal to the 

A/D converter, by using the appropriate gain settings in the 

pre-amplifier and the volume control. 

• The volume control should not generally be used to attenuate 

the analogue input signal. Any reduction in the signal level 

through the volume control may mean that clipping at the 

pre-amplifier output, due to excessive input signal level, will 

go undetected at the A/D input. 

The following analogue input control settings are 

recommended: 

Input signal type Typical signal Pre-amplifier Volume 

level 

setting 

Control setting 

Professional Line Level +20dBm 0dB 0dB 

Professional Line Level +14dBm 0dB 6dB 

Consumer Line Level +0dBm 20dB 0dB 

Dynamic Microphone -20dBm 40dB 0dB 

Audio Output Levels 

The digital output module in the system contains an 18-bit D/A 

converter and a volume control to provide the capability for 

driving analogue output levels less than the maximum 20dBm. 

The analogue outputs are electronically balanced. A schematic 

of the audio output circuit (on the D/A module) is shown below: 

Figure 107 — D/A output schematic 

The volume control settings on the D/A modules are 

manipulated by the upper row of sliders in the I/O Manager 

Tool. 

The following recommendations apply to the use of the D/A 

output module: 

• Always aim to supply a digital signal to the D/A converter 

that is close to full scale signal levels. If the digital output 

signal is scaled down significantly below full scale, it will 

lead to a reduction in the signal-to-noise performance of the 

D/A converter. 



• In general, the volume control should only be used to reduce 

the output level (-90 ... 0dB), and boosting the analogue 

output (+0.5dB … +30dB) is not recommended. This is 

because boosting the analogue output may cause clipping at 

the balanced line driver. 

• Ideally, the analogue output should be driven at the 

maximum possible level. For example, if the analogue 

output is intended to connect to a system that allows 

selectable gain on its input, it is always recommended that 

the lowest gain is selected at the receiving device and that 

the signal level generated by the output is increased. 

The following analogue output control settings are 

recommended: 

Output signal type Typical signal level D/A Volume Control 

required 

setting 

Professional Line Level +20dBm 0dB 

Professional Line Level +14dBm -6dB 

Consumer Line Level +0dBm -20dB 

Headphones driven directly by D/A -30 ... -10dBm (typical Adjust for comfortable level 

module 

Power amplifier driven from 

system[1] 

Note: If the power amplifier provides 

a volume control, refer to the next 

section for more information 

regarding the best settings for the D/A 

volume control and the amplifier gain. 

level) 

-30 ... -10dBm (typical 

level) 

Connecting Outputs to a Power Amplifier with Gain Control 

Adjust for comfortable level 

When an analogue output is connected to a power amplifier that 

also provides gain control, then the power level at the amplifier 

output can be increased or reduced by varying either the output 

volume, or the amplifier input volume. There are two common 

scenarios for this situation: 

1. In applications where the user wishes to control the volume 

of the amplifier output by adjusting the amplifier gain 

control. Optimum noise performance will be achieved by 

setting the analogue output level from the system to the 

standard line levels (0dBm for consumer equipment or 

+14/+20dBm for professional equipment), and then setting 

the amplifier gain control to set the desired listening level. 

2. In applications where the user wishes to control the level of 

the amplifier output from the Windows controls in VRack, 

then the power amplifier needs to be set to a suitable gain 

level. If the power amplifier gain is set too high, then the 

output level will need to be lowered to compensate, resulting 

in a situation where any noise at the outputs of the system or 

at the input to the amplifier will be accentuated. 


↓ To achieve the best results in 

this situation 


1. Reduce the amplifier gain to its minimum level. 

2. Boost the analogue output level to the appropriate line level 

(see the table above for the best setting for consumer or 

professional line levels). 

3. Increase the amplifier gain to set the loudest listening level 

that would be required during normal use of the system. 

4. Now reduce the analogue output volume to a comfortable 

level. 

The amplifier playback level can now be controlled using the 

D/A volume control in the I/O manager. 


Angle Channels 

Glossary 

Angle channels are utilised by Lake Technology’s simulation applications as channels for 

transferring the angle (azimuth, roll, and elevation) of a simulation object. 

Auralisation = Auralization: 

“Auralization is the process of rendering audible the sound field of a source in a space, in such a way 

as to simulate the binaural listening experience at a given position in the modelled space.” [Quoted 

from Kleiner, M., Dalenback, B., and Svensson, P. (1993), `Auralization-An Overview,'in Journal of 

the Audio Engineering Society, November 1993 , Page 869.] 

Auralisation and Acoustic Modelling Software 

Auralisation and Acoustic Modelling packages typically take a description of the geometry of the 

room (perhaps from a CAD package) and allow the user to specify the acoustic properties of the 

surfaces in the room, the properties and directionality of sources and also the properties and 

directionality of the listener room with the source and listener in the defined positions. 

Azimuth 

The angle from an observer to an object on a horizontal plane. Values start at 0.00 for forward and 

increase as the object moves towards the left, up to 360.00. 

B-Format Decoding 

Is the process of decoding B-format signals to a speaker array or headphones. 

B-Format Signals 

Are signals which are encoded in a three-dimensional audio format. The directionality of a sound 

field is represented by a spherical harmonic components, termed X,Y,Z and ω. The fundamental 

component is ω which is omnidirectional. X, Y, and Z are the first harmonic figure eight responses 

representing forward, left and up. 


Binaural HRTF 


Binaural Head Related Transfer Function, a pair of Head Related Transfer Functions, for both left 

and right ears. 

Convolution 

Convolution is a mathematical process that takes one signal and modifies it using another signal. It 

is equivalent to a multiplication of two signals in the frequency domain. This process allows one 

signal to be used as a filter for another. 

Crosstalk Cancellation 

During Transaural playback, sounds played back through a pair of speakers will overlap, creating 

crosstalk which can destroy the three-dimensional sound effect. Crosstalk cancellation provides a 

means of reducing this effect. 

DRAM 

Dynamic Random Access Memory. 

DSP 

DSP Stands for Digital Signal Processing and is a method for processing signals. Alternately, DSP 

stands for a Digital Signal Processor which is a computer chip designed to quickly process large 

amounts of mathematical calculations. 

Elevation 

The angle from an observer to an object on a vertical plane. Values start at 0.00 for forward and 

increase as the object moves upwards, up to 360.00. The elevation values may also run from -90 

(straight down) to +90 straight up. 

ERTF 

Environment Related Transfer Function. Similar to the HRTF however the ERTF represents the 

filtering of a sound source through the environment which it passes before reaching the HRTF 

component of an acoustic simulation system. 

FIR 

A Finite Impulse Response is an impulse response which after a period of time decays to zero. 


Order of Reflections 


The order of reflection is based on the number of surfaces a sound wave has reflected off before 

reaching the listener. Thus, first order reflections have bounced off a single surface, while third 

order reflections have bounced off three surfaces before reaching the listener. 

Frequency Domain 

A common way of referring to the mathematical representation of signals, where the signal is 

displayed in terms of its frequency components. The frequency domain representation of a signal is 

sometimes called a signal's spectrum. 

HeadTracking 

Is the real time process of monitoring the Angle, Elevation, Roll and Location of a listener’s head. 

HRTF 

Head Related Transfer Function, a mathematical model that characterises the filtering of a sound 

source before it reaches the eardrum. 

Impulse Response 

An impulse response is the way a system (such as a room) responds when presented with an impulse. 

An impulse response is specific to a particular system. The recorded impulse response of a system 

(such as those obtained using Lake Technology’s Measure tool) can be convolved with audio signals 

to create the impression of sound actually coming from the measured system. 

LakeNet 

Is the message protocol used by Lake applications for TCP/IP communication. External computers 

communicate with Huron applications using LakeNet. The LakeNet interface has been superseded 

by SNAP. 

LakeBus 

Provides a method for passing messages between Lake applications, along the Huron Bus. 

Latency 

Inherent delay due to signal processing. In Lake Technology applications latency is measured in 

samples or milliseconds. 


Linear Phase 


Linear phase is characterised by a constant time delay corresponding to a phase shift increasing 

linearly with frequency. 

Location Channels 

Are communication channels for transferring locational information of a Lake Technology AniScape 

or MultiScape simulation object. 

Low Latency 

Low latency is characterised by a short delay during signal processing. 

Optimum Phase 

By reducing optimum phase in Lake Technology’s EQ application, filters can be programmed to 

have zero or minimum phase delays. As a trade off for this low delay, phase distortion may occur. 

Reverberation 

The reflections of a sound source from the surfaces of a room or environment. 

RMS 

Root-mean-square amplitude of a sine wave. 

Roll 

The orientation of an object around a central point. Values start at 0.00 when the object is standing 

upright and increase as the object rotates clockwise, up to 360.00. 

Sample Rate 

Digital playback and recording is achieved by measuring the level of input signals at a specified 

sample rate. For instance at 48kHz 48,000 samples of data are taken every second. 

SNAP 

SNAP (Spatial Network Audio Protocol) is a protocol written to facilitate communication between 

an external application and Huron VRack applications. SNAP communicates with the Socket 

application using TCP/IP. 


SNR (Signal to Noise Ratio) 


The signal to noise ratio is defined as: The ratio in dB of the rms voltage to the rms voltage of the 

noise present. 

T60 

The time taken for a sound to drop (diffuse) 60 dB. 

TDM Channel 

A Time Division Multiplex Channel. 

Time Domain 

A common way of referring to the mathematical representation of signals, where the signal is 

displayed in terms of its components over time. 

Transaural 

Playback of a three-dimensional soundfield through a pair of speakers. Transaural playback often 

requires crosstalk cancellation to provide three-dimensional sound. 

XYZ Coordinates 

The XYZ coordinate system utilised by Lake Technology simulation tools represents the three axes 

of a three dimensional coordinate system. X and Y represent the coordinates on a horizontal plane, 

while the Z axis represents the vertical component of the system. 


VRack Autostart 

↓ To auto-start VRack at login 

↓ To automatically log into 

Windows 2000 

Appendix I: Auto-starting the Huron 

on power-up. 

The Huron may be configured to automatically run the VRack 

with a given workspace whenever the machine boots. The 

configuration is a two-stage process: 

• Configure the VRack to auto-start with a given workspace at 

user-login. 

• Configure Windows to auto-logon as that user. 

To auto-start the VRack at user-login requires adding a shortcut 

to VRack tool in Window’s taskbar Startup directory with 

desired workspace given as a command line parameter to the 

VRack tool. 

1. Right-click Properties on the taskbar 

2. Select the Start Menu Programs tab. 

3. Click on Advanced. 

4. Navigate to the Startup directory in the left hand pane, and 

add a new shortcut with a command line as: 

C:\huron32\Bin\rack32.exe myworkspace.wsp 

where “myworkspace.wsp” is replaced by workspace file 

that you wish to load on start-up. 

Test that it works by logging-out and logging-in again. The 

VRack should automatically start with your specified 

workspace. 

1. Download Microsoft’s Tweak UI tool (version 1.33) from 

http://www.microsoft.com/ntworkstation/downloads/powert 

oys/networking/nttweakui.asp. 

2. Install it (as administrator) by right-clicking on the 

tweakui.inf file that comes with the distribution. 

3. Run Tweak UI (as administrator) from the Control Panel. 

4. Select the Paranoia tab and ensure that Clear Last User at 

logon is disabled. 

5. Select the Logon tab, enable Log on automatically at 

system startup and enter the logon username and password 

you wish to use. 

6. Click the Apply button to make the changes take effect. 


↓ To specify an auto-logon 

domain 

THE APPENDIX 

If you’re login account uses the default login domain then that’s 

all you need do. Test that it works by logging out, you should 

find that Windows automatically logs you back in. 

However, if your login account uses a login domain other than 

the default then you will need to specify it manually by running 

regedit.exe as Tweak UI does not provide a means of doing this. 

1. Run regedit.exe and navigate to 

HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Wi 

ndows NT\CurrentVersion\Winlogon 

2. Modify the DefaultDomainName key by right-clicking it 

and entering your logon domain. 

3. Exit regedit and logout. You should find that Windows will 

now automatically log you back in again.

Huron & SNAP Documentation

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