sound analyzer 4.0 instructions manual

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Sound Analyzer 4.0 - Instructions manual

Acustica Applicata S.r.l. - Via Roma 79 - 55027 Gallicano - Lucca - ItalyTel. 0583 / 730322 - Fax. 0583 / 730914 - E-mail: [email protected]

Ufficio Tecnico di Varese: Tel. 0332 / 286562 - Fax. 0332 / 281253 - E-mail: [email protected]

Sound Analyzer 4.0Instructions manual

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Sound Analyzer 4.0Instructions manual

Sound Analyzer 4.0.....................................................................page 3Hardware and software requisites………..................................page 5Sound Analyzer setup - Quick start..........................................page 7Installing on Windows 3.1 / 3.11................................................page 9Installing on Windows 95 and Windows NT.............................page 10The “True RMS Level Recorder”...............................................page 11Level Recorder basic functions ……………………………........page 13Level Recorder advanced functions........................................page 18The “Audio Quality Test” signal...............................................page 27

The “articulation”of sound ………...………….................page 28 The “tonal balance” …………………………....................page 30

The “acoustic simmetry” of rooms ………..............................page 31The “A.Q.T. Analyzer” basic functions.....................................page 33A.Q.T. analyzer advanced functions …………………...............page 36Operating the Sound Analyzer on field.....................................page 40Performing the recording ………...............................................page 41Measuring the RT60 ………………………………........................page 41Performing an A.Q.T. measurement ........................................page 42Some examples of A.Q.T analysis ……….................................page 44Hardware and audio drivers troubleshooting guide ………....page 49Sound Analyzer technical specifications ………......................page 51

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Sound Analyzer is a software application which turns your PC into an accurate realtimeTRUE-RMS sound level recorder, with “A” and “C” frequency weighting curves and thetypical “slow”, “fast”, “reverb”, impulse integrating time constants.

It is also an automatic analyzer of the acoustic properties of listening rooms. Workingwith a custom derived audio test signal named Audio Quality Test (A. Q. T.), which hasbeen developed by Acustica Applicata after years of testing, it’s capable of taking“snapshots” of the acoustic behaviour of the “speakers-room” sound reproduction system.The analyzer creates simple graphs which can be compared and printed, and performsquantitative audio measurements that are in accordance with the impressions perceivedby the human ears.

The metering accuracy of the software is obtained by sampling the audio signal at 16bits (or more) and 44100 Hz (or more), and by means of a “software DSP engine”executing all its functions (TRUE-RMS level detection, “A” and “C” filtering and so on)with high-speed and high precision floating point calculations. The resolution and dynamicrange of level measurements is limited only by the accuracy of the audio card employedfor the A/D sampling process.

The Sound Analyzer “sampling engine” is based on the multimedia functions of Windows3.x or Windows 95 (Multimedia PC or MPC). Any Windows MPC compliant audio cardcan be employed (almost all of the sound cards for Windows now on the market). Agood audio card is highly raccomended, as the quality of the measurements is strictlydependent on the card’s low frequency response and signal to noise ratio. The accuracyobtained with high speed double precision floating point calculations is superior, ofabout one or two orders of magnitude, with respect to hardware detection approaches.

The software has no particular needs (of system memory, CPU speed or bus speed) inorder to work, and it does not take advantage of any non-orthodox programmingtechniques; for this reason it should work without any problem on any computer capableof producing and recording sounds using standard accessories of Windows, and withany audio card conforming to the MPC specifications. The need for high speed mathcalculation is satisfied by a Pentium 100 or better processor, anyhow suitable indicatorsof “realtime failures”, that is of “insufficient math speed”, are built-in.

The program has been tested with success on a large sample of audio cards, both“cheap” ones and those with a DSP (digital signal processor) on board, on ISA and PCIbuses, and also on notebook computers with an internal or PCMCIA 16 bit audio card.

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This is a partial list of tested systems:

- Creative SoundBlaster ( all models)- Turtle Beach (Multisound , Monterey, Pinnacle, Tropez + …)- Logitech SoundMan 16- Media Vision ProAudio Spectrum 16

- Diamond Aria 16- Diamond Monster Sound

- Audio System on portable computers based on chipset ESS Audio Drive- PCMCIA Wav Jammer card

- Crystal 423X chips

Sound Analyzer is a true “real-time” recorder, and so executes the sampling of audiosignals in real time; it also executes the detection of the RMS value of the signal, itsfiltering and its screen tracking in real time.

Because of these facts, it notably makes heavy use of both the audio sub-system andthe computer’s calculation resources.It is therefore essential that the most efficient and reliable audio card’s drivers areinstalled in the computer. It’s wise to use it alone, too, with no other programs running.As Windows 95’s multitasking is not a perfect one, it’s relatively easy to “stall” theexecution of the realtime calculations if interacting with the windows of other programs.This is note the case of the latest most powerful computers, with speedy CPUs and lotsof RAM.

A suitable indication of a “realtime failure” is furnished, anyhow, in the Sound Analyzerprogram. A “realtime” red indicator light will appear if, for any reason, the softwarethinks it has lost some samples, and that the measurement is not completely accurate.If this should happen you are advised to perform the measurement again, stopping theother softwares and maybe changing Sound Analyzer’s recording settings to lower itsneed of computing power, as described later in this manual.

The level recorder can be used to test the Rt60 (the standard decay time of the room),an important parameter of speech and music intellegibility.With the A.Q.T Analyzer window, and the use of the A.Q.T. test signal, you can performautomatic analyses of the room’s acoustical properties. These properties are directlyexpressed with numbers and quality measurement scales, so the comparison of differentacoustic setups is possible almost in realtime, and in a reproducible manner.

The main parameters measured by the A.Q.T. analyzer are the room “Articulation” and“Tonal Balance” (described later). These values are independently given for 4 differentfrequency regions starting from 20 Hz and ending at 1KHz. Average values spanningthe full frequency range are given too.

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Sound Analyzer 4.0 - Instructions manualHardware and software requisites

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The graphs (both in the level recorder and the A.Q.T. analyzer) can be zoomed andprinted. The A.Q.T. graphs can be compared (up to four at a time), so as to detect thedifferences between subsequent loudspeaker positions or between the left and rightchannels (to investigate for “audio image shifts - focusing”). The levels of details allowsto check the shape of even a single “frequency burst”, so as to detect the room behaviourin the attack, sustain and release of a tone.

Most of the information about the room behaviour, the echoes, the reflections and theirsum in phase or out of phase with the original sound, are all contained in an A.Q.T.graph. Sound Analyzer, in this sense, is an extremely accurate detector of listeningroom artifacts, and a real help to “ear only” calibration.

A hardware unit, named “Computer Controlled Sound Level Meter” by AcusticaApplicata, can be used as an input to your audio card as a preamplified microphone.This unit contains a software controlled variable gain high-fidelity microphone amplifier,and is furnished with a calibrated microphone with 20 Hz to 20 KHz frequency response+/- 2dB. This unit is directly controlled by the software.

You can also use a common good quality Sound Level Meter as a microphone for yourmeasurements. You loose the ability to perform the automatic audio input calibration,the selection of gain ranges, and maybe the frequency response is a little worse, butthe measurements will be good anyhow.

Hardware and Software requisites

There are NO special hardware or software requisites. The software works on a plainordinary Pentium PC, with an MPC compliant 16 bit audio card. Being highly mathintensive, Sound Analyzer benefits from a speedy CPU, expecially for the floating pointcalculation part.

Please note that for best performance and accuracy of the measurements the softwareuses only 16 bit sampling, so any older 8 bit audio card or subsystem won’t work.

No special requirements are there for the PC configuration. Here are some generalguidelines:

- CPU: Intel Pentium 100 or better (Sound Analyzer is a real-time recorder, anduses a lot of floating point calculations. So the more the math calculation power, thebetter)

- RAM: 16 MB or more

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Sound Analyzer 4.0 - Instructions manualHardware and software requisites

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- Hard Disk: 500 MB recommended (Sound Analyzer needs about 5Mb, morespace is required for the recordings and for the generated audio signals written aswave (.wav) files.

- Audio card: any 16 bit Windows MPC compliant (such as a Sound Blaster 16/32/64or compatible), half or full duplex (full duplex will enable you to use the built-in audiogenerator as a source while recording)

- The “Computer Controlled Sound Level Meter” by Acustica Applicata, or aSound Level Meter (Type I or II), with flat frequency response in the 20 Hz – 20 KHzregion.

Operating systems supported:- Windows 95/NT- MS-Dos ver. 6 or later with Windows 3.1/ 3.11 (not reccomended)

IMPORTANT NOTEs :

Sound Analyzer Software 4.0 is furnished on a set of 3.5" floppy disks or on a CD.

When used alone, without the “Computer Controlled SLM” by Acustica Applicata,the Sound Analyzer Software 4.0 is protected by a hardware key which must be installedon one parallel port (LPT1, LPT2 or LPT3 print connector). We remind you thatwithout this, the program will not work at all; it is necessary therefore to leave itinserted during both the installation and the use of the program. This “dongle” iscompletely “transparent”, therefore it can be inserted in series with any peripheral (suchas a printer).

When used with the “Computer Controlled SLM” by Acustica Applicata, the externalunit must be connected to a working serial port (COM1,COM2,COM3, COM4) via astraight serial cable. The recording functions won’t work if the link is broken or if the unitis not switched on. Please note that the use of the “Computer Controlled SLM” byAcustica Applicata, and the selection of the “com” port it is connected to must beaccurately configured at the first run in the “Recording setup” window (see LevelRecorder: Advanced Functions).

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Sound Analyzer 4.0 - Instructions manualSound analyzer setup

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Sound Analyzer setup

Quick Start

The Sound Analyzer software application can be installed by launching the “setup.exe”program on disk #1 or on the CD.

Remember to insert the hardware key , if necessary, or to connect the externalSLM unit via the serial cable and switch it on, otherwise the software won’t work!

Please follow the instructions on screen regarding the directory selection.

After the software is properly setup, you can launch the program by selecting the programname from the Windows menus (usually Start | programs | Sound Analyzer 4.0) , ordouble clicking its icon in the Windows Explorer.Sound Analyzer will boot and show its opening screen. After a few seconds you will betaken to the main (“True RMS Level Recorder”) window.

To maximize the precision of measurements, expecially at low frequencies, youare strongly advised to use the “line” input of your sound card, and not the“microphone” input. The last one usually suffers from a low frequency cut-off,and sometimes even an AGC (Automatic Gain Control) circuit is present, whichaffects the measured signal level.

Be extremely careful with the microphones built in on notebook computers; we haveexperienced that, on some of these units, the built-in microphone remains partiallysensible even after disabling its input (on the Windows mixer panel). Therefore, beforestarting your measurements, please check the exclusion of the microphone input andset its mixer bar to the minimum. If, even doing this, the mic is still partially active (yousee changes in the level recorded with all the other inputs muted), try to cover the smallopening of the microphone on the computer with a piece of adhesive tape.

These preliminary tests are very important for two reasons:

1°- The position of the internal computer microphone is not the ideal one for theacoustic measurements, and its frequency response or S/N ratio are certainly not Hi-Fi.2°- The recording could be disturbed by your eventual movements and noises on thekeyboard of the notebook.

Set up the microphone of your Sound Level Meter at the listening position by tilting theapparatus about 30° upwards from the horizontal level. If using a commercial SLM theunit must be internally configured to use the “C” weighting curve and the “Fast” integratingtime constant.

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Sound Analyzer 4.0 - Instructions manualSound analyzer setup

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Insert the AUDIO QUALITY TEST - ACUSTICA APPLICATA CD in your CD player,and start the A.Q.T. signal. If you have a Sound Level Meter with level reading (in dB),please set up the volume of your audio reproduction system so that the instrumentreads about 70 to 90 dB. If you don’t own one, please set the volume control so as toproduce at the measurement position a sound level well over the background noise,but not too loud (to avoid distortion). This level must be sufficient to excite the roommodes and the acoustical reflections on the walls we need to investigate.

After the set up operations, you will be ready to start the measurements. On the SoundAnalyzer the recordings are triggered by using the Go and Stop buttons (lower right onthe main window).

Please record a complete AQT signal (played back on your stereo equipment andpicked up by the Sound Level Meter and Sound Analyzer). Be careful to record accuratelythe signal without esternal interferences (such as background noise), including thestarting and ending reference tones, two short pulses of sinusoidal sound at exactly1000 Hz which occur about five seconds before and after the AQT itself. These tonesare essential to perform the automatic A.Q.T. analysis.

After the recording is complete (stop it after the second reference-tone by clicking the“stop” button”), you can analyze the measurement by pressing the A.Q.T. Analysisbutton (located on the right panel, about in the middle of the window).If the recording is OK and the reference-tones are properly recognized, a new window(“A.Q.T analysis”) will open, showing the analysis results. The A.Q.T. graph can nowbe zoomed or printed. New measurements and analyses can be performed withoutloosing the first one by selecting another measurement buffer in the first window (youhave four buffers to choose between).

This ends the quick start of the program. Anyhow, to exploit all of the advanced featuresand for a correct understanding of an A.Q.T. graph, we strongly advise you to read therelative sections in our manual, and to refer to the advanced textbook “Optimizing thespeakers-room system” by Acustica Applicata.

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Sound Analyzer 4.0 - Instructions manualInstalling on Windows 3.1 or 3.11

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Installing on Windows 3.1 or 3.11

Boot up Windows. Be careful you have no other programs running, both in DOS and inWindows (including the ToolBar of “Microsoft Office”). Shoot them down if necessary.

Insert the disk #1 in the floppy drive, or the CD in the reader.

Locate the setup.exe program on the installation media (disk or CD) with the FileManager and launch it with a double click. More simply you can select “Run” from the“File” menu in the program manager, and type the full path of this program (“a:\setup.exe”if the source drive is A).

The installation program will copy some temporary files, and ask you to insert the nextdisks if appropriate. During the reading of the second floppy, depending on the libraries(DLL) which are installed on your computer, it could be necessary to restart Windowsto allow some updates of the system files. If this should happen, you should resumethe installation from the first floppy by launching “setup.exe” again.

Please note that when Sound Analyzer 4.0 Setup’s second window appears, it’s possibleto accept the default directory for installation (namely “C :\ programs\ Sound Analyzer”)or to change it by selecting the “Change Directory” button.If the directory name is OK the installation can be completed by clicking the icon showinga monitor. The setup program will terminate the installation and will create the “SoundAnalyzer group” with the relative icon of the program.

It is now wise to exit and restart Windows to allow for the changes to complete.

When loading the program, the image of next page will appear. Please make sure thatyour audio card name is properly displayed in the “recording section” in the lower rightside of the window, just over the “Go” and “Stop” buttons. If not please change the“setup” of the parameters of the program (see the description of the “File|Recordingsetup” function).

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Sound Analyzer 4.0 - Instructions manualInstalling on Windows 95 and Windows NT

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Installing on Windows 95 and Windows NT

Boot up the operating system. You will need to have administrator privileges if youhave Windows NT, otherwise you will not be allowed to install system components.Please refer to your system administrator.

To launch the setup.exe program please press the “Start” button, then select the“Run” menu label and when a small window opens please type “a :\setup.exe” andpress the “return” key. Alternatively you can browse the floppy #1 or the CD (dependingon your installation media) and launch “setup.exe” by double clicking on it.

Please take care that no other programs are running, both in DOS and in Windows(Including the ToolBar of “Microsoft Office”). Shut them down if necessary.

The setup program will copy some temporary files. While reading the #2 floppy youcould be prompted to restart Windows to allow for same updates in the system filesarea. If this should happen, please allow for the reboot, then resume the installationfrom start (by launching “setup.exe” again).

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Sound Analyzer 4.0 - Instructions manualThe “True RMS Level Recorder”

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Please note that when Sound Analyzer 4.0 Setup’s second window appears, it’s possibleto accept the default directory for installation (namely “C :\ programs\ Sound Analyzer”)or to change it by selecting the “Change Directory” button.If the directory name is OK the installation can be completed by clicking the icon showinga monitor. The setup program will terminate the installation and will create the “SoundAnalyzer group” with the relative icon of the program.

It is now wise to exit and restart Windows to allow for the changes to complete.

When loading the program, the image of the previous page will appear. Please makesure that your audio card name is properly displayed in the “recording section” in thelower right side of the window, just over the “Go” and “Stop” buttons. If not pleasechange the “setup” of the parameters of the program (see the description of the“File|Recording setup” function)

The “True RMS Level Recorder”

The “True-rms Level Recorder” window is the main interface for audio measurements.It shows a graph on the left containing the “virtual paper” of the level recorder, and asuitable set of icons (on the right and at the bottom of the window) to setup themeasurements parameters and perform the recording.

Up to four measurements can be executed or loaded from disk in each session. Thefour “measurement buffers” can be switched by selecting one of the four round buttonsat right, just below the “Acustica Applicata” logo. Only one buffer is active and displayedat a time. The actions selected on the relative button at right (the one with three dots“ …” ) work on the active buffer. When clicking on this button a pop-down menu willappear, showing options to clear, load or save the measurement, and with other optionsdetailed below.

In the lower right corner of the window you can see the recording triggers. The “Go” and“Stop” buttons control the start and stop of the recording. On their left there are twowarning indicators which will become red in case of “realtime errors” (the computer wasnot able to keep the pace of the recording with the math calculations) and of istantaneous“input overload”.

While recording the “overload” indicator changes its state according to the actual levelin input. An “overload memory indicator“ ( a little “M”) will remind you that the input hasoverloaded during some part of the last recording session.

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Sound Analyzer 4.0 - Instructions manualThe “True RMS Level Recorder”

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The time lenght of the measurement, and the percentage of how much of the recordingbuffer is filled are shown at left.

A drop-down list, on the immediate left, will enable you to setup the measurement withthe appropriate RMS “integrating time constant”, that is the number of millisecondsused to integrate each of the RMS values of the signal (100 per second are recorded).The standard “Slow”, “Fast” and “Reverb.” time constants are allowed, together withtwo new values: “A.Q.T.”, “Impulse”. The “A.Q.T.” setting, which emulates the humanear time constant of about 25 milliseconds, will be used in the A.Q.T. analysis.

Again at left there is a drop-down list of available filter curves. The “Type A” and “TypeC” settings allows you to use the standard frequency weighting curves usually found oncommercial Sound Level meters. The actual use of the filter during the recording sectionis enabled by putting a check mark in the check-box just over the drop down list.

Just below the buffer selection buttons you can see a lot of icons. These buttons controlmost of the functionalities of the recorder. A button with and “audio card” icon allowsthe audio card selection and calibration, as illustrated below. An icon with a CD allowsyou to enable a simple “CD player controller” (which will appear on the lower left corner).This space is shared by the “audio wave player”, which is triggered by the icon with asinusoid on it. This is a simple wave files player which can be used instead of the CDplayer as a source of measurement signals.Of course simultaneous recording and playing will be possible only if you have a properlyconfigured “full duplex” audio card.

After a recording session, or after loading a measurement from disk, you can perform aset of analysis on the signal. The eight buttons just over the recording section allowsyou to zoom and pan on the “virtual paper” of the graph; the icon with a printer allowsvarious kinds of prints; the “Rt 60” panel (in the middle of the lower panel) will enableyou to make accurate measurements of decaying time constants based on yourmeasurements.

And last, if you have correctly recorderd an “A.Q.T.” signal, you can use the “AnalyzeAQT” button to perform an automatic measurement of the acoustic properties of yourlistening environment.

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Sound Analyzer 4.0 - Instructions manualLevel Recorder basic functions

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Level Recorder basic functions

The menu commands in the Level recorder were kept to a minimum in order to realizea visual interface which resembles the one of a phisical measurement instrument. Mostcommands are given by selecting an icon, a check-box and so on, so as to emulate theuse of knobs and switches. Only some functions, namely program setup, printerselection, and Audio Generator selection are performed with commands enbedded inthe classical Windows menus.Each command and icon action will be now accurately described.

(“File” menu)

Info- Shows the program logo with informations on software licence and program version.To close the window, click with the mouse on it.

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Recording Setup- Opens the “Recording Setup” window, which allows the configuration of variousprogram parameters (see “Advanced functions” later in this manual).

Printer Setup- Opens the “printer setup” window, which allows for the selection and setup of theprinter that you want to use to print the graphs.

Print- Opens the “print” window, allowing the choice of the type of print (see “Advancedfunctions” later in this manual)

Exit- Terminates the Sound Analyzer program.

(“Recording” menu)

Start Recording- Starts the recording. The samples will be enqueued into the active buffer.Stop Recording- Stops the recording. The recording will be automatically stopped if the audio buffer isfull.

(“Audio generator” menu)

Generate Test Signal- Opens the “Signal generator window” which enables you to create “wave” files (.wavfile extension) containing the A.Q.T test signals (see “Advanced functions” later in thismanual).

(“A.Q.T. Window” Menu)- Takes you directly to the A.Q.T. window

(“Print” icon)Opens a small window which allows the selection of print type andprints graphs of the signal. Equivalent to the “Print” menu voice ofthe File Menu.

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(“Program info” icon)Opens the “Program logo” window, containing informations aboutprogram licence and software version. To close the window clickwith the mouse on it. The same as the “Info” function of the FileMenu.

(“Terminate program” icon)Terminates the Sound Analyzer program. Equivalent to the “Exit”function of the the File Menu.

(“Go” button)Starts the recording. Equivalent to the “Start Recording” function ofthe Recording Menu.

(“Stop” button)Stops the recording. Equivalent to the “Stop Recording” functionof the Recording Menu.

(“A.Q.T. window” button)Takes you directly to the A.Q.T. window. Equivalent to the“A.Q.T. Window” menu function.

(“…” or “buffer functions” button)The buttons with three dots open a pop-down menu with thefollowing functions:

- Clear Audio Measurement: Clears the measurement buffer(the recording).- Load Audio Measurement: Loads an existing measurementfrom a file (with a “.ame” extension) in the active buffer. SaveAudio Measurement: Saves the measurement in the activebuffer on a file, with a “.ame” (Audio measurement) extension.- Import Wave File: Allows to import a 16 bit Wave File,acquired with a different application, into the active buffer.The wave file can be mono or stereo (a suitable channelselection window will appear), and must be recorded at 16 bitand 44100 Hz.

- Analyze A.Q.T.: Executes and automatic A.Q.T analysis on the signal contained inthe active buffer. If the analysis is possible (starting and ending tones correctlyidentified, sufficient Signal to Noise ratio and so on), it opens the A.Q.T. window anddraws the new A.Q.T. graph.

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- Remarks: Opens a window containing a little notebook. You can write some notesabout the measurement on it, such as the recording setup or operating conditions.These notes can be later printed along with the graphs (see “Advanced functions”later in this manual).

Please remember that up to four measurements can be kept in memory simultaneously,but only one measurement buffer is active at a time. Only the correspondent “…”commands button will be enabled too. To operate on a different buffer you must firstselect it as active by selecting the option button at left.

(“Analyze A.Q.T.” icon)“Analyze AQT” performs the AQT analysis on the acquired(or loaded) signal and activates the A.Q.T. window.

(“Overload” warning light)When the input signal saturates the input stages of the A/D converter,the signal is “clipped” and an accurate measurement of its RMS valueis not possible. This warning light will turn red while the input signal istoo high to be accepted by the audio card input. The overload is amomentary condition, so the light will change from red to gray assoon as the signal level drops to confortable values.

(“Realtime” failure warning light)This warning light will turn red (and stay red) if there has been a “realtime failure” during acquisition, that is if some samples were lost, andthe measurement is not fully accurate. The indicator will reset whenthe next recording will start.

(“Overload memory” indicator)“M” - Overload memory. The “overload” is an instantaneous condition,so this indicator remains on when there has been a momentaneousclip even if now the signal range is within the allowed limits. Theindicator will be cleared when a new recording is started.

(“Buffer status” indicators)Acquisition buffer status: shows the actual recording time and howmuch of the buffer capacity is actually employed.

(“RMS Integrating time” drop down list)You can select one of the following choices:- Slow: 1 s. , the standard SLOW setting of Sound Level Meters- Fast: 125 ms. , the standard FAST setting of Sound Level Meters- A.Q.T. : 25 ms. , used to emulate the human ear time constant.

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This is the default setting for A.Q.T. analysis- Reverb: 16 ms. , the standard REVERB setting on the most accurate Sound Level

Meters- Impulse: 1 ms., used to reveal even the smallest changes in level. Useful for RT60

measurements.

(“Input filters selection” panel)Allows you to insert the filter (when the check-box is marked) and tochoose its frequency response between:- “Subsonic” : cuts the frequencies below 2.5 Hz in order to removeany DC level- “Type A” : the standard ANSI S1.42-1986 type “A” filter.- “Type C” : the standard ANSI S1.42-1986 type “C” filter.

(“Zoom” and “Pan” icons)Icons that allow the zooming and panning of the signal.From left to right and from top to bottom:− pan left− zoom on the X (time) axis− unzoom on the X (time) axis− pan right− pan down− zoom on the Y (level) axis− unzoom on the Y (level) axis− pan up

The pan or zoom effect is “amplified” if you keep the “shift” key pressed on the keyboardwhen clicking on the icons. A “Pan – all” or “zoom-max” can be obtained by keeping the“shift” + “ctrl” keys depressed. The same applies for “unzoom” operations.

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Sound Analyzer 4.0 - Instructions manualLevel recorder advanced functions

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Level Recorder Advanced functions

(“File | Recording setup” menu function)Opens the “recording setup” window, where you can adapt some of the program

characteristics to the performances of yourcomputer, and to enable the use of theautomatic computerized SLM by AcusticaApplicata.

Recording setup section:

Show realtime level bar (Default Settingactivated):enables the realtime level bar (right of thegraph) during the recordings. The bar usesCPU resources, so it can be disabled in orderto avoid “realtime” failures.

Fast Screen update (Default Setting activated):activates a faster redraw method for the signal graph. The new method is lessaccurate, and requires a complete redraw when the recording is stopped, butuses less CPU resources and can help to avoid the “realtime” failures on slowercomputers.

Recording buffer (Default Setting 2 MB):sets the size of the internal sampling buffers. On fast computers you can use theSMALLEST value, freeing RAM memory for other applications; on the slowestones you can try to increase the buffer to compensate for the lack of processorspeed, in order to avoid “realtime” failures. This is particularly useful with CPUsrunning below 120 Mhz.

IMPORTANT NOTE:For CPUs below 120 Mhz (expecially on notebooks, which are naturally slowerthan desktop machines) and with less than 16 Mbit of RAM, we advise you to tryto disable the realtime level bar function, to use the “fast screen update” functionand to enable a larger “recording buffer” if you occasionally get “realtime” failuresduring the measurements. The software is highly CPU intensive, expecially whenthe weighting filters are used during the recordings. These settings try to lowerthe load on the CPU by disabling the less useful features, or by compensatingthe lack of speed with larger buffers.

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Sound Level Meter section:

Use Sound Level Meter (default setting: disabled)When checked it enables the use of the Computer Controlled SLM by AcusticaApplicata. This unit must be connected via a straight serial cable to the serialport connector selected. The recording functions won’t work if the unit is notproperly recognized.When the external unit is recognized at program startup, a new panel will appear(where there is usually the RT60 box) that allows you to control its settings. Withthe new buttons you can select:

− the gain range of the amplified measurement microphone− the use of the “calibration tone” as an input to the audiocard, in order to calibrate exactly the measured levels.The unit will be automatically controlled when performing the “inputcalibration”

Com port (default setting: “COM 1”)Allows you to select the correct serial port the SLM is connected to. You canchoose between the 4 ports usually seen by Windows (COM 1, COM2, COM3,COM4).

(“Print” menu function or icon)Opens the window shown below, in which you can choose 4different types of graph:Visible window – prints the full visible window displayed on themonitor

Visible signal on X strips – prints only thevisible portion of the signal graph on X strips(1 or 2 per page).Full signal on X strips – prints the full signalon X strips (1 or 2 per page).Full signal with exact scale - prints the fullsignal with exact scaling (in mm/s), so as toemulate the paper on a real level recorder.You can select between 0,1 / 0,3 / 1 / 3 / 10/ 30 / 100 mm/s as a scale.You can select the Color Print if you own acolor printer.You can select the Print Remarks Page ifyou want to print a page with the remarksabout your measurements.

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(“Generate test signal” menu function)Opens the “A.Q.T. Signal Generator” window shown below.You can generate the A.Q.T signal (and write it on disk into a WAVE file, 16 bit , 44100Hz, mono or stereo). The kind of signal can be chosen between:

− mono− stereo, AQT on LEFT− stereo, AQT on RIGHT− stereo, AQT on bothchannels.

The different signals are usefulfor the various measurements inthe room, to excite only one orboth channels of your stereoequipment.The A.Q.T. signals areacoustically marked with two

reference tones (5 seconds before and after the signal itself). The frequency of thereference tone (Ref.Tone freq.(Hz)) is selectable between various values ( 1000 Hz,892 Hz, 924 Hz, 956 Hz, 1116 Hz ….). Different frequencies are useful to avoidusing a tone at a frequency whose level is distorted by your room’s resonances. Thesystem could not be able to detect the tones accurately in this case, so you can selecta different frequency, possibly that of the “cleanest tone” on the graph.The “Generate” button will open the “Wave File name” window, allowing you to choosethe name of the file. Usually the test signals are saved in the “wave sub-directory “ (thatis “\Sound Analyzer\wav”, where they can be quickly located with the “Wave Filegenerator”.

(“RT60 measurement” icon and panel)When the RT60 icon is depressed, a line with two cursors will be

shown on the graph (see the figure onthe left), and a panel with a white boxcontaining the measurement value willappear at the bottom of the window,overlapping the zone where the SLM controls areeventually located.

The measurement of the RT60 time (the time, in seconds,employed by the sound level in the room to decay to 1/1.000.000 of the original level, or –60 dB), requires youto select an appropriate region of the graph containing adecay of the sound level. You will never have 60 dB ofdecay in your graph, of course, but please ensure that a

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sufficient S/N value is present (at least 30 to 40 dB). The correct decay time is calculatedfrom the slope of a line that must be “overimposed” as accurately as possible on thedecay zone of the graph.

You should position the first cursor over the top part of the decaying signal, and thesecond one where the level has decayed to noise, trying to match the slope of thegraph with the slope of the line segment between the two pointers. The measurementof the reverb time will appear in the RT60 white box, expressed in seconds.Press the RT60 icon again to stop the function and make the SLM controls reappear.

(“Tone markers position” icon)By clicking on the icon with an arrow pointing upwards, youwill enable the manual selection of the position of the two

reference tones. Two arrow shaped cursors will appear, one labeled “start” and onelabeled “end”. They can be positioned in correspondence with the reference tones,before and after the test signal.

Usually the program automatically analyzes the A.Q.T. signal by detecting its positionbetween the two reference tones. Sometimes undesired impulsive noises are presentbefore or after the tones themselves, preventing the program to recognize the beginningand the end of the A.Q.T. signal. In this case an error window opens showing themessage ”Warning: low (starting or ending) reference tone S/N !”.You can try to correct the spikes with the “pencil” icon function described at next point,this is always the first thing to try. But if the automatic detection of tones is still difficult,you can manually position the tone markers on them. Be extremely careful in thisoperation, the detection and optional “streching” of the A.Q.T. depends on the accuracyof tone positions.

These “spikes” can be corrected , with the “pencil” function by substituting the undesiredspikes with a small horizontal segment which should line up with the level of thesurrounding noise. How is this done? Let’s observe the images in the following page:the first shows an impulsive disturbance, a few seconds before the initial referencetone.

(“Signal eraser/corrector” icon)This “signal eraser/corrector” icon activates the erasing functionof undesired spikes. These spikes are particularly dangerousin proximity with the two reference tones, as they interfere

with the detection of the correct start and stop of the measurement. The spikes can be“leveled off” to the level of the background noise by selecting the “pencil” icon.

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Undesiredspike

The mouse cursor will turn to a pencil, and when the left mouse button is kept depressedthe points under its tip will be adjusted to the desired level. You can, in other words, “re-paint” the signal where desired. Please be careful not to erase the A.Q.T. signal or thetones themselves.So try first to “zoom” the image on the X axis to see the spikes in detail. Then press the“pencil” icon, position the cursor a few millimeters before the spike (with the Y level setto the surrounding noise level), click the left mouse button, and keep it depressed whileyou drag horizontally the cursor to a few millimeters after the undesired disturbance.Release the mouse button, and the graph will be redrawn. Repeat the operation for theothers spikes next to the two reference tones.

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(“Audio card setup” icon)

This icon opens the “Input Device Setup” window, shown below. Itis used to select the audio card (in case you have more than oneoutput on your computer), and the sampling parameters.

You can select between 3 audio sampling formats:-44.1 Khz, 16 bit-48 Khz, 16 bit-96 Khz, 24 bit

Only the audio formats supported by your audio card will be actually enabled, the otheroptions will be grayed.

(“Audio Input Calibration” button)

The calibration of the line input with a suitable signal is necessary to display the correctabsolute level values (in dB) as the Y scale on the graphs. These scales represent theabsolute audio level of the sound picked up in your environment.

All audio levelmeasurements are referredto minimum sound levelpressure (20 microPascal)taken as a reference. Thisis the average earingthreshold of our ears, and isindicated as the absolute 0(zero) level in dB. All theother sound pressure levelsare indicated as dB(decibels) over the reference0. A level of 70 to 100 dB istypical, for instance, whenlistening to music atconfortable levels.

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Please note that, unlike your ears, your audio card doesn’t come with a reference levelbuilt-in. Moreover the “mixer bar” allows us to change the gain of the audio input, that ishow much the signal we are feeding in is amplified. So no absolute level measurementis possible, only differences can be measured.

To have the Sound Analyzer to display the correct sound pressure level in dB, we mustfirst preset the mixer bar to a certain position (and leave it unchanged thereafter). Thisis a first “gross calibration” of the input. Then we must feed the input with a knownstable level that we know is generated by the amplified mic when a certain SPL ispresent, and have the software to compensate for the differences in audio cards. Inother words we need a “reference level calibrator” to preset the internal calculations sothat the exact SPL is displayed when the mic feeds a certain voltage level into the audioinput.

By selecting the “Audio Input Calibration” button you can perform the calibration of yourcard input level. This is a three step process:1) you connect the “refefence level generator” to the line input, and preset the mixer

bar so as to avoid “overload” warnings;2) you write the “conventional SPL” associated to the calibration source in the

appropriate box;3) you let the software calculate its internal compensating constants.

If you own the “Computer Controlled Sound Level Meter” by Acustica Applicata,the calibration process is almost automatic. This unit contains a “reference levelgenerator” which produces a low distortion sinewave at about 1000 Hz, 1 Vpeak. If youclick the “Audio Input Calibration” button, the Sound Analyzer will automatically tell theexternal unit to feed this “reference tone” as an input, and will automatically read its“conventional dB level”. All you need to do is to preset the mixer bar an press the“calibrate” button.

The calibration is simple, anyhow, even if you don’t own this unit, but you have a SLM.Most good quality Sound Level Meters have an internal “calibration tone” for calibratingpurposes. If you have one of these units you can use this tone (and its “conventional dBlevel”) to calibrate the software.

As a final resort, if your Sound Level Meter doesn’t include a reference tone, you canuse its level measurement ability. Use a stable sound from a test CD (a sinewave forinstance, or “white noise”) to produce a stable sound level of about 90 to 100 dB in yourroom. Start the sound and write down the level displayed by your instrument. Write this

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number in the “calibration level” white box when asked by the wizard. While the soundis still playing (at a stable level, of course), preset the mixer bar to avoid the input stagesaturation and press the “calibrate” button.

The calibration is guided by a self-explaining wizard. Follow the instructions displayed,and click the “Next” button to advance to the next step. Please write the “referencelevel” in dB in the white box if asked to (if you own the “Computer Controlled SoundLevel Meter” by Acustica Applicata this step will be skipped). Set the audio mixerbar to the highest possible level without overload, then press the “Calibrate” button.After this calibration takes effect, without moving the mixer bar anymore, the dB valueshown by your Sound Level Meter, must match with the value shown by the SoundAnalyzer. The last window will show you a Finish button. Please note that the calibrationcan be aborted by selecting the Cancel button.

Please remember that the calibration is necessary to obtain measurements with accuratelevel scales (in dB). The calibration must be performed at least once when you firstinstall the software, and whenever something changes in the measurement setup (youchange the audio card, or the mixer levels). A calibration is useful even from time totime, to compensate for the temperature drift or component aging.

After calibrating the reference level, please check the displayed levels, if possible, withthe external SLM. If differences bigger than 1 or 2 dB manifest, than perform thecalibration again. This check is not necessary if you own the Acustica Applicata“Computer SLM”.When checking the measured levels against the ones displayed by an external SLM,please note that 1 or 2 dB of difference is quite common because of the differences inthe mic positions or directions.

(“CD generator” icon)Toggles the “CD Generator “ panel, in the lower left corner of the window.This panel is used to to control the CD player of your computer. It allowsyou to select the desired CD track, and to start and stop the playingfunction. The track elapsed time is displayed when the track isreproduced.

(“Audio wave generator” icon)Toggles the “Audio Wave Generator “ panel, used to load a signal froma Wave file and use the PC as an audio generator. You must first selectthe wave file to play (mono or stereo, but at 16 bit, 44100 Hz) by clickingthe button at left, then, when the file name is correctly displayed at thebottom of the panel, the playing controls will be enabled.

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You can use this built-in generator to play the A.Q.T. signals generated by the “A.Q.T.generator” function, or any other wave file you have (maybe with a “white noise” orother test signals).The wave file will be played will be played on the first available wave output device(usually device #0). If you own a “full duplex” audio card you will be able to use the levelrecorder while the wave file is played; otherwise you will get a warning message windowsaying that the computer resource (the audio subsystem) is actually busy.

(“Computer SLM” control panel)

If you have a Computer Controlled Sound Level Meter byAcustica Applicata (hereafter indicated as “Computer SLM”)properly connected and setup, a new panel will appear mid-wayin the bottom of the Level Meter window. This panel contains 5buttons: three of them (the ones on the lower line) are used to

set the input gain of the microphone preamplifier, that is to change the “full scale range”of your meter, while the others two on the first line are used to choose between the mic(“Mic” button) and the built in “reference calibrator” (“Cal” button).

When the Computer SLM is enabled in the “Recording Setup” window (as describedbefore), the Sound Analyzer will try to initialize the external unit at startup. If the link isOK, and the unit is switched on, the control panel will appear, otherwise a warningmessage will be shown and no Computer SLM controls will be available.

These buttons are enabled all the time, even when you are recoding a sound level. Thiswill allow you to adjust the desired full scale level and to choose the type of input signalfor test purposes. The panel will be covered by the “RT60” panel when this advancedfunction is shown.

The “Audio Quality Test” signal

The A.Q.T. is a “multitone” test signal which has been developed by Acustica Applicatato perform an advanced analysis of the acoustic properties of the speakers-room system,with the capabilities and limitations of the human earing system in mind.

It can be played directly in the listening room to take a “snapshot” the behaviour of thereproduction system and its interaction with the room. It consists of a sequence ofsinusoidal tones, rising in frequency with time, intermixed with short periods of silence.It can be employed to listen to your listening environment and be able to detect the

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change in its behaviour when the tone’s frequency changes; the different “dynamics” atthe various frequencies will be clear, and also any strange “resonance” effect excitedby the sound in the room.The Sound Analyzer software advances the analysis power giving you automaticallyboth a graph of the listening system‘s behaviour, and quantitative measurements of thedynamics of sound as seen by our ears for both the full frequency range and for fourdistinct frequency regions:

- at low frequencies (from 20 to 50 Hz) – the “Lower Bass” zone- at upper bass frequencies (from 52 to 100 Hz) – the “Upper Bass” zone- in the warm zone (100-300 Hz) – the “Warm Zone”- at lower mid frequencies (from 300 to 1000 Hz) – the “Lower Mid” zone.

The “multiburst” signal was developed by Acustica Applicata after years of intensivetesting on field. The tones start from 20 Hz and rise up to 300 Hz with a step of 2 Hz,from 300 Hz to 1000 Hz with a step of 4 Hz, and from 1000 Hz to 2000 Hz with alogarithmic step. This change of “frequency resolution” has proved to be the bestcompromise between frequency resolution (with respect to the human earing capabilities)and lenght of the test.

Each “single frequency tone” lasts 3/15 of a second (200 ms.), and is followed by 1/15(66ms) of silence. So each complete “burst” lasts for 4/15 s. , and the full signal spansa total of 94.13 seconds (not including the starting and ending reference tones, whichare used for automatic start and end detection purposes).

The structure of the signal is not casual, of course. The A.Q.T. is the closest possibleto a musical signal and, like that, it has frequencies that change with time, and changesof sound level, intermixed with small pauses of silence. Each tone, like an impulsivesound from a drum, increases its level, remains at level for some time, then decays.

In 3/15 of a second of tone, we have the possibility to investigate the sound of thesurrounding, in 1/15 of a second of silence, we verify what we find in the “reverberated”field, that is all that portion of sound which comes to our ears not directly from thespeakers, but from the walls of the room where we listen.

The A.Q.T. signal is digitally generated by the program itself ( with the “AudioGenerator” function). It’s written on disk as a 16 bit per sample wave file. In thissense it’s as perfect as the 16 bits of resolution allow for.

The dynamics of the A.Q.T. in a wave file, or on the test CD, is the maximum allowed bythe actual digital-audio standard. The signal is mathematically constructed and thereforeit is perfect both in its form and duration. Each “burst” is made with a pure sinusoidal

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tone at a prescribed frequency lasting 3/15 s., with smoothed starts and ends, followedby 1/15 s. of silence. To be exact there is not exactly silence, that is series of “0”s ondisk, but a “1 bit dither” so as to avoid problems with certain CD players which mutetheir DACs when they see “0”s in input.

We use 6 dB of margin to avoid overloads in the output stages of CD players, so thesignal is recorded at –6 dB relative to full scale. The signal on disk has about 90 dB ofdynamics.

When recorded directly by connecting the output of the CD player to the audio card, orwhen imported directly from the Wave file, the A.Q.T. signal will mantain much of itsdynamics. When properly zoomed, each tone will look almost perfect, with straight linesegments in the attack (almost vertical), sustain (horizontal) and release (vertical again,with negative slope) portions. Please remember that we are recording the “true rms”level of sound, so we have an almost stable level when the sinusoid is playing, and thelevel of noise in correspondence to the silence zones.

When recorded with a mic at the listening position, the signal dynamics is usually bemuch lower. The reflections and room modes dramatically limit the decay of sound inthe room. So what you see in the A.Q.T. graph is the dynamics of your speaker-roomsystem” versus frequency. In this graphs you will notice that sometimes the tones arestill almost perfect (and the music, at the same frequency is perfectly intellegible),sometimes their shapes is completely changed. In the worst cases the structure of thetone can be almost completely lost: the room contributions have overwhelmed the originalsound.

The A.Q.T. test is an optimum one to pick out the best position for the speakersand the listening position. It’s extremely useful to perform an accurate and correctacoustic tratment to the room too.

The “articulation” of sound

You may have noticed, after importing the wave file containing the A.Q.T. signal, thatdespite you see high level of dynamics in the graph (maybe 80 to 90 dB), the “articulation”values calculated by the program are much smaller.

This is not an error, since “articulation” is not sound dynamics.

When evaluating the quality of an audio reproduction system, traditionalmeasurements are useful, but don’t tell you the full story. They usually don’t takeinto account the properties of your ears.

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The human ear is a complex apparatus whose behaviour is actually only partially known.Some of its properties have been investigated into deep; one of the most important isits “masking behaviour”, both in frequency and time.

When the human ear is excited by a sound, it looses some of its “resolution” withfrequency: some frequencies, those that are near to the exciting sound, are“masked”. This “masking” remains for some time even when the original sounddisappears. It’s like we have got a “frequency mask” with a shape of a bell,centered on the exciting frequency, which slowly decays to zero with time.

The important fact: in 1/15 s. (66 ms.) the masking of the ear decays about 20 to25 dB.

So only the reflections that have decayed less than 20 - 25 dB can be actually heardafter the period of silence. If the sound level in our room goes down , in the silenceperiod, more than this limit, then our ears will get all the sound’s details they can.

Please note that it’s not “the more, the better” in the listening environments.

If the decay time is extremely small, and the level drops down to – 30 or –40 dB (withrespect to the sustain level), maybe the room is too “dead”, and the sound is not thebest one, you lack the contribution of room reflections, that are “natural” to your ear.If the decay time is high, and only 5 to 10 dB of dynamics is present at a certainfrequency, the room is too “live”, maybe there are even echoes, the images shifts towardsthe reflections, and some details of the sound are lost under them.

So here is the “articulation”:

it’s the dynamics of the room as seen by our ears.

The Sound Analyzer is actually the only audio measurement system which is capableto correlate the physical measurements to the actual sensation at the human ear. Workingon the shape of tone decays, it’s able to calculate numerical values that are directlyconnected to our listening sensations. In accordance with the behaviour of our ears,the articulation values calculated when using the A.Q.T. signal have a max of about 26dB.

Our experience on field has proved that an improvement in perceived soundquality is always reflected in the increase of the measured “articulation” values.So these are the numbers to look at when optimizing the speakers-room system.

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The “articulation” is a frequency related number, that is you can express an articulationvalue for each separate tone burst. This value is shown both in the drop-down menu(triggered by right-clicking on the graph) and when the “single burst analysis” window isdisplayed.

The “articulation ” graph globally shows the articulation at each frequency as a linerepresenting its value, in dB, above 0. In this kind of measurement, of course, the morethe better, but a good average level is even more important.The Sound Analyzer displays average values for the four frequency zones illustratedbefore, and an average value for the full 20 Hz – 1000 Hz range.

Please note that only average values are important: it’s not important to have the bestvalues at certain frequencies if you have the worst ones at the others; it’s better to havea good average articulation level.The Sound Analyzer automatically calculates average values for the four frequencyzones shown before, plus a “global” mean value spanning the frequency range from 20Hz to 1000 Hz.

The “tonal balance”

Another important quality parameter of your listening environment is the “tonal balance”.This is practically a sort of “frequency response” of your system, but it’s a morespecialized measurement as only the “sustain average levels” are taken into account inthis case.

The purpose of the “tonal balance” measurement is twofold:1. to test for the equilibrium of “timbre” of your listening environment.2. to check for the differences between the two stereo channels, which are the

main causes for the shift of the audio images and for the lack of “focusing” ofaudio sources.

The “tonal balance” graph consists of a line representing the difference, in dB, of thefrequency tones level with respect to the level of the 1000 Hz tone (taken as a reference).In this kind of measurement, of course, the less the shift the better.The Sound Analyzer displays average values for the four frequency zones illustratedbefore, and an average value for the full 20 Hz – 1000 Hz range.

Please note that our aim, in this case, is to obtain the lower average shift for each

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zone, and the lower global shift too. Big changes limited to a few frequencies are notcritical if we have a low average shift value.

The “acoustic simmetry” of rooms

Given the Sound Analyzer’s ability to compare two or more signals on the screen, wecan in general verify the graphic resemblance of two subsequent recordings. If weplay the A.Q.T. only one channel at a time on our stereo reproduction system, we caneven get a “snapshot” of the “acoustic simmetry” of our room. We can even investigatethe possibility to have a “well focused sound” ( that is a stable central stereo image) atall the frequencies .

The human audio perception system (ears + brain) is capable to locate a virtual soundsource in space mainly using level differences (between the ears) and delay of soundarrivals. If the two channels of our audio reproduction system play a certain tone at thesame level, the source is located in the middle; if one channel is louder than the other,the source shifts towards that speaker. If the difference in level reaches 20 dB, thesource is actually located in the louder speaker.

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So if we compare the tonal balance of the two stereo channels analyzed separately, wehave a “snapshot” of the “image shift” or “lack of focusing” problem:1. If the difference in level between the two channels is high at a certain frequency, the

image will shift towards the louder channel.2. If the difference is reversed at other frequencies, the shift will be reversed too.3. If we have a sound source with a complex spectrum , the source will “broaden”: the

lower frequencies will shift in one direction, the higher ones in the other; we will getthe impression of a source of sound ( a voice, an instrument) without “natural”dimentions.

Please note that “articulation” and “tonal balance” are often strictly related: large changesin level (tonal balance) are generated by the interaction (in phase or out of phase) ofdirect and reflected sounds. If we have a dip or a hill in the tonal balance graph, maybewe have also a strong reflection which interacts with the direct sound.

The “A.Q.T. analyzer”

The window of the “A.Q.T. analyzer” is similar to that of the level recorder, but with a lotof differences. Four independent buffers are available in the A.Q.T window too, butunlike the ones in the level recorder window, they are all displayed at the same time. Soon the right side of the A.Q.T. window we have the four buffers (initially labeled “1”, “2”,“3”, “4”) with check-boxes, and not option buttons, to enable or disable each of them.

You can simultaneously display and compare up to 4 A.Q.T. graphs side by side. Thisis useful to compare the changes in your room response with various levels of acoustictreatment, or with changes in loudspeaker positioning. Whenever you change somethingin your stereo setup you can record the A.Q.T. signal in a free buffer and analyze it,comparing the differences with the previous situation (the graphs before and after thechange).The A.Q.T. graphs can be saved and loaded from disk (with a “.aqt” file extension), anda set of functions is available for them too. These functions are displayed in a pop-down menu triggered by clicking on the “three dot” button at right.

Just below this panel you can see a drop down list with three options:“A.Q.T.”, “Articulation” and “Tonal Balance”. You can select the typeof measurement information to display. The first choice displaysthe A.Q.T. graph, that is a “level vs. frequency” graph, with eachsinusoidal tone of the signal properly centered ad its frequency.

The “Articulation” and “Tonal Balance” graphs, on the contrary, show two “global” analysisparameters, namely the ”articulation level” at various frequencies and the “tonal balance”of the room response. More on this later.

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When displaying the graphs some numerical values appear in the lower part of thewindow. Here you can see the average values of “articulation” or “tonal balance”calculated for the four typical frequency regions (namely “lower bass”, “upper bass”,“warm zone” and “lower mid”).

A set of icons appears just below the drop down list. Some of them are similar to theones in the first window, and trigger the same functions (such as print, zoom, pan andso on). Some icons are new, with specialized functions for the A.Q.T. analysis. Theirmeaning will be described in the following.

A.Q.T analyzer basic functions

The visual interface is again icon-based. Most commands are given by selecting anicon, a check-box and so on, so as to emulate the use of knobs and switches onmeasurement instruments. Only some functions performed with commands embeddedin the usual Windows menus. Each command and icon action will be described accuratelyin the following sections.

(“File” menu)

Printer Setup - Opens the “printer setup” window, which allows for the selectionand setup of the printer that you want to use to print the graphs.Print - Opens the “print” window, allowing the choice of the type of print (seethe “Advanced functions” paragraph for the Level Recorder)

Close A.Q.T. Windows - Closes the A.Q.T. window, bringing you back to the main(level recorder) window.

(Right button drop-down menu)

When moving your mouse cursor on the graph background, try to click the right mousebutton and keep it depressed. A drop-down menu will appear, with some usefulinformations and functions.

Frequency - Displays the frequency of the point in the graph under your cursor tip.

[1 .. 4]Level / Articulation - Shows the actual average level (in dB) and meanarticulation of the selected frequency tone for the four A.Q.T. buffers. This is useful tocompare the values at certain frequencies in order to understand if some improvementswere obtained between the measurement sessions.

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Align average levels - Allows you to level off the average level of the graphs. Thisis useful to overlap them just in case the measurements were taken at different listeninglevels. This works by automatically introducing the right offset value into the A.Q.T.signals.

Clear Offsets - Clears the offsets of all of the graphs.

Single Burst analysis - see “Advanced functions” later in this chapter.

(“Buffer enable/disable” check box)Up to four A.Q.T. analysis can be simultaneously displayed on one graph. Use thesecheck boxes to enable or disable each buffer.

(“…” or “buffer functions” button)

A set of functions is appliable to each buffer containing theA.Q.T. These functions are displayed in a pop-down menutriggered by clicking on the “three dots” button at right:

- Clear A.Q.T. : Clears the buffer, and redraws the display.- Load A.Q.T. : Loads an A.Q.T. analysis from disk (A.Q.T.sare saved with a “.aqt” file extension).- Save A.Q.T.: Saves the actual A.Q.T. analysis to disk (withthe “.aqt” file extension).- Trim A.Q.T. Position: Opens the “A.Q.T. position Trim”window (shown here at right). You can use this function toshift accurately the signal withrespect to the frequency scale (tothe left or to the right) in order toposition the tones exactly betweenthe frequency tickers; the upward

arrow brings the signal back to the Default position.

- A.Q.T. Offset: Allows you to set the signal’s vertical offset; after the analysisthe signal has 0 (zero) dB offset, but you can change it from -60 dB to +40 dB (in 10 dB steps) to avoid the overlap with different signals (for display or printpurposes, or to better compare their differences). There is also a “custom” offsetfunction, allowing you to input whatever offset value you desire (maybe to overlapexactly two graphs).- Remarks: Opens the “Audio Measurement Remarks” window, a small notebookwhere you can write your notes that will be saved with the measurement andprinted, if you want, with the graphs.

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- Change room parameters – Opens a window where you can change thedefault room dimentions and its measured reverberation time. These values areused to calculate the “room modes bar” which can be overlapped on the graph.

Please note that themodes are referred to asingle A.Q.T.measurement, asdifferent measurementscan be loaded whichwere taken in differentrooms. Set the desiredvalues and press OK.The values will be savedin the correspondent file.See below for details.

- Line color: It is possible to change the 4 A.Q.T. default colors.

(“Print” icon”)Opens the print window. Equivalent to the “Print” function of the File Menu.

(“Close A.Q.T.” icon)Closes the A.Q.T. window, and takes you back to the level recorder.Equivalent to “Close A.Q.T Window” in the File Menu.

(“Zoom” and “Pan” icons)Allows the zooming and panning of the graph. They work just likethe ones on the main (level recorder) window.

(“Default room parameters” icon)Activates the default room parameters window. If you are performingrepeatitive measurements in the same room, once its dimensionsare set, these are automatically assigned to every recording.

(“Graph type ” drop-down list)Allows you to select the graph type between three types:- A.Q.T. – The acquired multiburst signal with level andfrequency scales

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- Articulation – Displays a line representing the “tonearticulation” of the signal (see later for a description of the term).- Tonal Balance – The same for the “Tonal Balance” (seelater in this chapter)

(“Analysis results” panel)The numeric values of the average “articulation” of the four analyzed signals are displayedin this panel at the bottom of the window. From left to right we have the mean values forthe “Lower bass” (20-50 Hz), the “Upper bass” (52-100 Hz), the “Warm zone” (102-300Hz), and the “Lower mid” (302-1000 Hz) zones. At right the values with the yellowbackground are the total average articulations of the entire signal. When the “TonalBalance Analysis” graph is displayed, they show the same values for the Tonal Balance.The last four boxes on the right show the name of the file. See later for the understandingof these terms.

A.Q.T analyzer advanced functions

(“Single Burst analysis” menu function)This advanced analysis function is available in the drop-down menu (triggered by right-clicking on the graph). When selected, it will show the “Single Burst Analysis” window,which enables you to analyze directly each frequency burst, and to compare the valuesfor the four loaded measurements. The graphs and the values displayed can also beprinted. For each burst (a period of sinusoidal signal plus a period of silence) the followingvalues are calculated and shown:

- Frequency : the tone frequency- Starting overshoot: the level of overshoot in the raising part of the tone; lower

values are better, they mean better room sound control.- Average level : the average level for the “sustain” part of the tone- Max deviation : how much the level oscillates around the average level in the

“sustain” zone. The lower the better, of course, as the level in this part of thetone should be stable.

- Ending overshoot : how much the level overshoots before the decay. The lowerthe better

- Articulation : the calculated “articulation” of the tone. See later for an explanationof the term. The higher the better in this case.

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You can skip to the adjacent tones with the “Previous burst” and “Next burst” buttons.This allows you to move over the A.Q.T. signal.

The “RT60“ panel and button, in the top part of the window, allows you to calculate the“RT60 at the selected frequency”.

This is an advanced option which is useful to detect the problems of resonances inrooms. The decay time is not constant at all frequencies; it’s easy to understand thatwhen the frequency excites a resonance mode of the room, the sound will decay slower.The RT60 calculated in the “level recorder” window is the max decay tima with all thefrequency range excited, so it represents the “slowest” possible room decay: the situationis certainly better at some frequencies. This can be investigated within the “Single burstanalysis” window. You will probably see that the “cleanest” tones have also the smallestRT60.

(“Wavelenght info” icon)By pressing this button, a small window will open over the graph showing the frequency,the relative wavelenght and half-wavelenght, of the point under the tip of your mouse

cursor. The window is permanently displayed on top of the other windowswhile visible. These informations are useful to analyze the A.Q.T. graph,when you try to match some graph properties (dips or hills) to some physicalcause. It’s useful to point at some zone on the graph and know exactly thefrequency, the wavelenght and half-wavelenght .. Is there an “in phase”sum of direct and reflected sound, for instance?

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(“Room modes” icon)By pressing this button, the program will calculate the main resonances of the room (its“room modes”), and their successive harmonics. The calculated values are displayedas a colorful bar at the bottom of the graph. Short segments horizontally lined on threelevels and with three different colors are displayed, one for each room dimension (“W”

for Width, “H” for Height and “L” for Lenght).

In general there’s a simple mathematical rule to calculate the modefrequencies:

fn = n * 344 / (2*l)where “l” is the room side lenght in meters and “344” is the speed of sound in metersper second.The values drawn on the graph are referred to the room sizes of the selected referenceA.Q.T. measurement. This reference graph number is selected in the drop down list atright. The room parameters, in fact, are relative to each measurement, and are mantainedin its “.aqt” file. These values can be changed for each A.Q.T as explained before.Please pay attention to select the appropriate “reference” buffer if you want thecalculations to be based on its room parameters.

You can see that different background colored areas are present too. Their meaning isexplained in the following.

Important note: Frequency areas in axial modes

Different areas of different colours are used as backgrounds for axial modes. Thesecolours are used to give you a level of confidence in the possibility that some measuredeffects (in the graphs) are actually originated by the room modes.

The audible frequencies have notable variations when viewed in terms of wavelenght.At 20 Hz we have a wave of 17,2 meters, while at 20 KHz a full wave is just 1,72 cm.The analysis of sound behaviour in enclosed spaces varies from the low frequencies,practically omnidirectional and influenced very little by small obstacles, to the higherfrequencies which, having smaller wavelenghts, behave like a billiard ball, arecharacterized by geometrical reflections and trajectory changes and are susceptible tosignificant modifications depending on the surfaces they meet on their course.Therefore, for a more accurate analysis, we divide the audible spectrum in four frequencyareas:

- Area A (white on the graphic) - below the fundamental resonance mode of the longerroom side. No acoustic reinforcement due to room resonsnces is possible at thesefrequencies, as the half-wavelenghts at these frequencies are longer than the longestroom side. Only reflections are important in this area.

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- Area B (yellow on the graphic) - this starts from the lower modal resonance andarrives at a threshold which cannot be exactly defined, because the shift to the successivearea is very gradual. Some acoustic researchers have approximated the borderfrequency (F2) with the equation:

F2 = 1875 RT60 V

where 1875 is a costant, RT60 is the calculated decay value in the room, and V is thevolume of the room.In this area we can properly talk about “room modes of resonance”. Here we will seethe widest effects caused by the modal resonances of the room.

- Area C (light blue on the graphic) – is limited at left by the F2 value defined above,and at right by F3 = 4xF2.

It is the most difficult zone to analyze, since the waves at these frequencies are tooshort to be investigated by wave acoustics (as those in the B area) and too long to bethought as ‘rays’, with geometrical behaviour similar to that of light. In this area diffusionand diffraction effects energetically model the sound wave that reaches our ear; it isimportant to consider both the level of absorbtion and the shape of wall surfaces and ofall the other obstacles met by the the sound while in travels toward us.The difficulty of the analysis, and the importance of this zone on the final listeningresult, are made evident by the fact that in rooms of ‘domestic’ sizes, we are already inthe range of maximum sensibility frequencies of the ear, and we find ourselves immersedin sounds which comprehend both speech (voice spectrum 150-4000 Hz approx.) andmusic ( 30-10.000 Hz approx.).

- Area D (white on the graphic) - from F3 to the highest audible frequencies. We canconsider the behaviour of these frequencies as “geometric”, with a propagation whichis very directive and with rebounds off the obstacles which respect the laws concerningthe rays of light: angle of reflection = angle of incidence.

Please note that a direct match between graph artifacts and room modes isplausible only in the B area, while the effect at higher frequencies is probablymuch smaller. Keep this in mind when correlating the effect to the possible causes.Be advised. Moreover, that many effects are overimposed on the A.Q.T. graph,so a direct correlation with a single problem of the room will exist only in rareoccasions.

√

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Sound Analyzer 4.0 - Instructions manualOperating The Sound Analyzer on field

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Operating the Sound Analyzer on field

How to prepare for the measurement

To execute correct measurements the software must be installed and properly working,and the audio input must be calibrated with care, both for the input mixer settings andthe final calibration with the reference level. Please perform these steps before goingon with the measurement.

Please try to record some samples of sound at the desired measurement level . Duringthe signal acquisition, make sure that the mixer of your sound card do not overload theaudio card’s input stage. We strongly advise you to use the line input of your card, andnot the microphone input. The last one is usually heavily filtered, especially at lowfrequencies, and often uses an automatic gain control (AGC) circuitry that prevents thesystem from measuring correctly the RMS value of the signal.Moreover be extremely careful with notebook computers; we have experienced that onsome of these units, even disabling its input in the Windows mixer, the microphoneremains partially sensible. Therefore, before starting a recording, please verify theexclusion of the microphone input, and lower its bar on the mixer to the minimum. If,even doing this, the mic remains partially active, try to cover the small opening of themicrophone on the computer with a piece of adhesive tape.

Position your Sound Level Meter in correspondence with the listening point, and tilt theapparatus about 30° upwards from the horizontal level.If you use a Sound Level Meter as your mic, this unit should be configured with the ‘C’or “linear” weighting curve (for a flat frequency response), and with the “fast” integratingtime constant. The “Computer Controlled Sound Level Meter” by AcusticaApplicata, on the contrary, does not require any setting, as the filtering and integratingfunctions are realized in software.

On the Sound Analyzer please select the “A.Q.T.” integrating time constant, and NOinput filter.If you suspect low fequency noises (or wind, for instance), you can select the “subsonic”filter, but be advised that the filters use a lot of computer resources, so use the only ifnecessary. The ‘C’ and ‘A’ filters can be employed for the usual noise SPLmeasurements, like the ones on the others SLMs. They won’t be used for the A.Q.T.analyses, because no low frequency filtering must be present when measuring the lowfrequency behaviour of your speakers-room system.

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Sound Analyzer 4.0 - Instructions manualPerforming the recording and RT60 measurements

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Insert the CD with the track containing the A.Q.T. signal in your player. While adjustingthe mixer levels of your sound card, if you have a Sound Level Meter with level readout,set the volume of the reproduction system so that the instrument reads about 80 to 100dB. This level is sufficient to excite the room modes and the other acoustic phenomenawe need to investigate.

Performing the recording

Your computer and the Sound Analyzer are now ready to work.Clear the measurement buffer if necessary.Start the sound source.Click on the “Go“ button, and check that no overload happens while you use yourreproduction system to play the signal you wish to record.

Please remember that the Sound Analyzer software can be protected by the means ofa hardware key to be installed on one parallel port (LPT1, LPT2 or LPT3 printerconnector). We remind you that without this, the program will not be able torecord.The same applies if the software is employed with the Computer SLM by AcusticaApplicata. No hardware key is present in this case, but the Computer SLM must becorrectly working and be connected to your computer serial port if you want the “record”function to be avaliable.

Performing RT60 measurements

If you want to measure a reverberation time, generate a white noise at a level of 80/90dB for about 10 seconds, then suddenly stop the reproduction and wait while the soundlevel in the room returns to that of background noise.Stop the acquisition using the “stop“ button, then press the “RT60” button: two cursorswill appear, with a line segment connecting them. A panel labelled “RT60” with a whitebox will appear mid-way at the bottom of the window.The cursors must be positioned with the mouse so as the line overlaps the slope of thelevel decay. For optimum positioning of the line segment, we suggest you to expandthe acquired signal horizontally using the relative keys.

The RT60 value can be read in the white box in the “RT60” panel. Please note thateven if the acquired signal does not reach 60 dB in decay, the program will anywaycalculate the correct reverberation time giving a RT60 value.

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Sound Analyzer 4.0 - Instructions manualPerforming an A.Q.T. measurement

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Performing an A.Q.T. measurement

To perform an automatic A.Q.T. analysis, you must record the level of the sound whileplaying this signal, including the starting and ending reference tones. Please note thatthe maximum aquisition time for each buffer of the Sound Analyzer is is 150 s. , so abuffer will be almost filled by the A.Q.T signal test (about 110 seconds).

It’s advisable to generate the A.Q.T. with a level of about 70 to 90 dB; when the soundis playing the bottom of the graph must be well over the background noise level (thelevel before the first tone), and you must check for “overload” or “realtime” errors.

When the recording is finished, click on the “Analyze A.Q.T.” button, and wait until theprogram processes the acquired signal and shows the “A.Q.T. Analysis” window.

You can manually position the “Reference Tones positioning arrows” if necessary. If,after clicking on the the “Analyze A.Q.T.” button , an error window opens saying ” Warning:Low (starting or ending) reference tone S/N”, the reference tones cannot be accuratelydetected, so you can manually mark their position with the cursors. In this case wethink it’s wiser to repeat the measurement, increasing the audio level just a bit. Themeasurement will be more accurate.

The same problem can be caused by undesired noise “spikes” near the tones. You can“level off” the spikes with the “pencil” function, as illustrated before. We suggest you torepeat the measurement, anyway, as the spikes could have corrupted also the centralpart of the graph.

Once the signal is processed, let’s pass to its interpretation. Try “zooming” the signal. Ifyou notice that the tones are not correctly “centered” between the frequency slots, try touse the “Trim A.Q.T. Position” function to shift the signal. Remember that each frequencytone starts from background noise, raises to the desired level (with or without a startingovershoot), stays leveled for a few milliseconds, then drops to noise level again (or tothe reverberating level of the room). So the start of the tone is where the slope of thelevel starts to raise. This is in theory, of course, as reverberation will prevent the previoustone level to drop to zero. The “silence” period (1/15 s. , or 66 ms.) is shorter than theperiod of signal presence. The room should decay as fast as possible in this “sourcesilence” time. Take a look at the articulation values, and look for higher values whenchanging your listening setup.

Lets imagine making successive measures in a room where we are realising an acoustictreatment ; lets load into buffer N°1 the A.Q.T. signal we have recorded in the roombefore the start of the acoustic treatment, into buffer N°2 the A.Q.T. signal after the firststep of acoustic treatment.

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By selecting the “Articulation” graph type on the screen, two coloured lines (one foreach buffer) will be shown on the graph. Each line oscillates between 0 dB and somemax value (10, 20 ..). If you compare the shape of the two lines, you will notice that the“post-treatment” one reaches higher values at certain frequencies. The difference inlevel between the two lines represents, at the various frequencies, how many dBs youhave increased the articulation (dynamics) of the room examined. You have a numericalmeasurement of the improvement caused by the acoustic treatment in the room. Thisshould reflect also in the average values shown at the bottom of the window.

This comparison can be used with up to 4 different signals simultaneously. So manydifferent alternative setups can be tested. Please remember that an acoustic treatmentis always a critical one, and that sometimes you get an improvement at certainfrequencies and things get worse at others. Numbers and line shapes under your eyesare a big help, anyhow, expecially with acoustical phenomena. With Sound Analyzerand a little experience “on field”, you can wisely try to improve any acoustic problem.

Our experience shows that, even in acoustically corrected rooms, it’s practicallyimpossible to obtain more than 10 dB of articulation under 50 Hz. It is very difficultto obtain more than 12 dB between 50 and 100 Hz. It’s possible, instead, toobtain higher of articulation above 100Hz.

This behaviour is in accordance with the low frequency behaviour of sound in enclosedspaces: the lower the frequency, the longer the wavelenght, and the longer the decaytimes too. In general very low absorption is possible at low frequencies, and evenacoustic absorbent materials loose their properties when frequency goes down.

The casuistry of who has used our methods of acoustic investigation, demonstratesthat a huge part of the listening problems are due to the “warm zone” (102-300 Hz)behaviour, and to the first portion of the “lower mid” zone (300-500 Hz). Increasing thearticulation and the tonal balance in this range of frequencies, you can achieve anoticeable sound improvement.

These are only introductive notes. For a better undestanding of the A.Q.T. propertiesand uses we advise you to check our reference text : “Optimizing the speakers-room system”.

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Sound Analyzer 4.0 - Instructions manualSome examples of A.Q.T. analysis

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Some examples of A.Q.T. analysis.

A.Q.T recorded in the same room, before and after the acoustic treatment. In thefirst case ( in the upper part of the screen), we observe that the dynamics iscompressed (average articulation is 6,1 dB) and that the frequency response(tonal balance) shows wide differences even for adjacent frequencies.

In the second case, after an accurate acoustic treatment, we observe an increaseof the articulation (now an average 20,9 dB) and a total line-up of the tonal balance.

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The same A.Q.T. graphs shown in the previous page, rightly “zoomed”, allowyou to analyze in detail the zones of maximum interest.

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A further enlargement shows the multiburst structure of the A.Q.T. signal and themultiple distortions introduced by the room acoustics.

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The “Single Burst Analysis” window allows you to improve the acousticinvestigation to see how each single tone “burst” is changed by the room. Hereare depicted all the values of acoustic interest. The RT60 calculation can beperformed too.

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This graphic shows two AQT signal recordings for both the left and the rightchannel, seperately. The lines represent the tonal balance of the two channels,and show a distinct acoustic asymmetry since the two lines in different zones arealso distanced by 10 dB. At those frequencies the image, originally recorded tobe reproduced in the middle between the two speakers, will shift half-way betweenthe middle position and the louder speaker.Especially in the areas where the two lines are more distant, a correct focus ofthe acoustic signal will not be present.

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Sound Analyzer 4.0 - Instructions manualHardware and audio drivers troubleshooting guide

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Hardware and audio drivers troubleshooting guide

After a “clean” installation, the software should work without any problem if the computerhas working multimedia audio extensions. In our esperience serious problems, like“system crashes “ or “Windows general protection errors“ (which cause the program tocrash or even Windows to exit) were only generated by incorrect audio drivers installedon the computers, or when these drivers were not sufficiently stable, or when theaudio subsystem was not properly configured (its resources, such as the I/O addressrange, or DMA channel, or Interrupt vector number, were in conflict with those of othercards or other software programs).

In general you should ensure to have the audio subsystem of Windows correctly workingbefore installing the Sound Analyzer. If the Windows multimedia apps are working, at16 bit and 44100 Hz, Sound Analyzer will work too.

Anyhow the following notes suggest some techniques for problems resolution.

1)Make sure that your audio card doesn’t have conflicts with other hardware installed.Check the I/O addresses of the card, its Interrupt numbers, and DMA channel use.Please refer to a technician if you don’t understand these terms.Keep in mind that if the computer works with the Windows Recorder it would also workwith the Sound Analyzer.If you use Windows 95/98 it is possible to check for any presence of conflicts with thecomputer “control panel”.If there are no conflicts go on to next step.

2)Make sure that your audio card’s drivers are correct. Please ensure that the drivers youhave are suited for your operating system. If you bought your card some time ago,maybe before the Windows 95 release, the drivers on its disks are probably 16 bitversions, and they won’t work or won’t install correctly on Windows 95/ Windows 98.The same considerations applies if you run Windows 98, which uses a different drivermodel. Maybe you need updated drivers. Keep in mind that old drivers always meanlower performances, if they work at all.

3)Always use the most updated audio drivers. Often cards builders distribute new releasesof drivers by BBS or on a WEB site on Internet. These new releases are needed tocorrect functional errors and to improve the efficiency. Our experience tells us that if theprogram is recording but something is still wrong with the recordings (strange spikes, or

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Sound Analyzer 4.0 - Instructions manualHardware and audio drivers troubleshooting guide

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unfrequent lock-ups), in 99% of the cases the problem is caused by the stability of thedriver, so if you have access to a new release, your problems can be solved.On Internet the new audio drivers for your card can be found at the card’s factory WEBsites. You should probably find it on the instruction manual of the card. If not we adviseyou to check some WEB sites which contain collection of links to manufacturer’s sites.Please try at “http://www. conitech.com/windows” or “http://ftp.coast.net/Coast/”.

4)Make sure that no other drivers are installed into your computer which take advantageof your audio card, or “monitor” its inputs (to check for incoming calls, for instance).There are multifunctional audio cards, with DSPs on board, which can be used likeFAX/MODEM, either for FAX reception or for vocal synthesis or analysis, or like anelectronic secretary. These is possible even with the new ISDN adapters.These drivers CAN cause conflicts with the use of the sound card as a sampler, so itmight be worth trying to disable or even remove them. If the audio is working but youget errors such as “the specified resource is allocated” when starting a recording, thisshould be the cause.Keep in mind, anyhow, that the audio quality of these combined devices is often scarce.Maybe it should be appropriate to install a good quality audio-only card to make goodquality measurements.

5)This general statement remains true, anyway: if the Windows recorder works, the SoundAnalyzer MUST work.

6)In the unfortunate case that, after all the checks, you still have serious troubles with theSound Analyzer program, please notify us accurately about your problems and we’ll tryto help you.We ask you to first gather all the possible informations regarding the type andconfiguration of the computer (type of processor, RAM memory size, bus type ecc.), ofthe sound card (including interrupt numbers, I/O addresses and DMA channels), of theoperating system, and, if possible, detailed informations on the installed drivers for thecard.If the problem is with a notebook computer please try to identify the “audio chip”’s nameand number. Try at the “control panel”, and write all the informations you find regardingthe audio card and its drivers.

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Sound Analyzer 4.0 - Instructions manualTechnical specifications

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Technical specifications

Level Recorder:

− High speed sampling using an MPC compliant 16 bit, 44100 Hz or better,audio card

− Software True RMS detector, 64 bit floating point, dynamics limited only bythe A/D process resolution (16 bits or more).

− 100 integrated samples per second.− 4 independent recording buffers with max 15.000 points (150 s. max recording

time)− 5 integrating time constants: Slow (1 s.), Fast (125 ms.), A.Q.T. (25 ms.),

Reverb (16 ms.), Impulse (1 ms.)− 3 linear phase software filters: “A”, “C”, “Subsonic”. Filtering can be excluded.

Filtering performed in software via realtime 1024 point FFTs, 64 bit floatingpoint calculations.

− Overload indicator with memory.− “Realtime failure” indicator.− Audio card level calibration wizard.− RT60 measurement function.− Automatic rec stop on buffer full.− Graph zoom and pan functions.− Graph print with 4 print options.

A.Q.T. Analyzer

− 4 independent buffers simultaneously on screen for comparison purposes.− Automatic analysis of the A.Q.T. signal, with automatic start and stop detection.− 3 types of display: A.Q.T, Articulation, Tonal Balance− Automatic calculation of average Articulation and Tonal balance for the full

A.Q.T. frequency range, and for the 4 frequency zones: Lower Bass (20 Hz –50 Hz), Upper Bass (52 Hz – 100 Hz), Warm Zone (104 Hz – 300 Hz) , LowerMid (304 Hz - 1000 Hz).

− Advanced “Single Burst analysis” capabilities.− Graph zoom and pan functions.− Graph print with 4 print options.− Overlaid room modes display.− Interactive frequency and wavelenght info window.

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Sound Analyzer 4.0 - Instructions manualTechnical specifications

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A.Q.T. (Audio Quality Test) signal

− Digitally generated “multiburst” signal.− Peak level: -6 dB re full range.− 90 dB dynamics range.− Smoothed sinusoid start and stop.− Dithered “0” level to avoid muting functions of A/D or D/A converters .− Each burst or “single frequency tone” lasts 3/15 of a second (200 ms.) and is

separated from the next tone by 1/15 (66ms) of silence (4/15 s. total).− Tones starts from 20 Hz and rise up to 300 Hz with step of 2 Hz, from 300 Hz

to 1000 Hz with step of 4 Hz, and from 1000 Hz to 2000 Hz with a logarithmicstep.

− Start and end reference tones (1000 Hz default, variable frequency), -6 dBrelative to clipping, for automatic detection purposes.

− “Signal generator” function to generate the signal on 16 bit, 44100 Hz wavefiles on mono files, and on stereo files on Left, on Right, and on Left + Rightchannels.

sound analyzer 4.0 instructions manual

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