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Smaart 7 Impulse Response Measurement and Analysis Guide

1. 3 OG D 50 250 450 650 850 105k 125k 145k 1 65k 185k 205k 225k 245k 2 65k Result The spectrum of the selected time range is displayed in the Frequency graph e BE E e ae kl 80 Lin Y Ze F heater way a a BS E E Geesse 40 aw 80 F Log 50 250 450 650 850 1 05k 125k 1 45k 1 65k 1 85k 205k 2 25k 245k 2 65k Frequency Smoothing 1 6 Oct 30 50 70 100 200 300 500 700 2k 3k 5k Yk 10k Figure 12 Moving the time O point and selecting a time range for display The time range selected in the navigation pane applies to the Frequency graph as well as time domain graphs Lin Log or ETC Note that Smaart uses a tapered data window when transforming any subset of the full IR time range We ve drawn the outline of a Hann window in red on the navigation pane of the result portion of the illustration above to help visualize this The Spectrograph Smaart s Spectrograph straddles the time and frequency domains giving you a birds eye view of frequency domain characteristics of the IR over time If you are familiar with Smaart s real time spectrograph the IR mode version is essentially the same display rotated 90 If not one way to think about how it works is to start with the idea of a spectrum analyzer On a real time spectrum analyzer RTA you typically have a bar graph or line chart with frequency on the x axis and magnitude in dB on the y axis showing you the spectrum of some chunk of
2. See www sengpielaudio com calculator RT60Coeff htm Air absorption Air absorbs sound quite significantly at higher frequencies due to molecular resonance This high frequency air absorption varies with temperature and relative humidity in a way too complex to cover here However a useful air absorption calculator may be found at the link below www doctorproaudio com doctor calculadores en htm calc humid 53 Page Copyright 2015 Rational Acoustics LLC All rights reserved 0 3dB m 0 2dB m 6kHz 0 1dB m 3kHz 5kHz f 0 0dB m O 10 20 30 40 50 60 70 80 90 oefficient Relative humidity Absorption Air absorption in dB m vs RH at room temperature Note Sound absorption through air is per meter not per doubling of distance as seen for radial attenuation Sound in a room with a mid band RT60 of say 2 seconds will travel several hundred meters before decaying into the noise floor so air absorption is a significant additional factor in reducing high frequency reverberation times Schroeder cut off frequency RT60 assessments are usually restricted to the Schroeder region of the frequency range where mid and high frequency wavelengths are short compared to the room dimensions and sound paths become dense diffuse and fairly random At lower frequencies individual modes occur These modes are more related to room dimensions less dense and don t follow the diffuse reverberation characteristics required for a
3. The effective directivity of an acoustically small source tends to be governed by local boundaries The illustration shows a large room with some identical acoustically small red spherical sources in various positions with respect to the room boundaries Ignore Victor He s only there to avoid 3 D ambiguity e The source that is dangling in free space has a Directivity Factor Q of 1 and radiates its acoustic power spherically into full space e The mid floor source Q 2 has its acoustic power concentrated into a hemisphere It radiates the Same power but concentrated into half space e The baseboard source Q 4 has its acoustic power concentrated into a quarter sphere Again it radiates the same power but this time concentrated into quarter space e And the rear corner source Q 8 has its acoustic power concentrated into an eighth sphere radiating the same power but this time concentrated into eighth space Useful on axis free field sound pressure level headroom increases with respect to Q 1 will approach 6dB per boundary 6dB for Q 2 half space 12dB for Q 4 quarter space 18dB for Q 8 eighth space Microphones near boundaries Similar spl increases apply to pressure microphones if they are placed acoustically close to a surface i e much less than 1 6 of the shortest wavelength of interest At 1 6 wavelength the round trip will be 1 3 wavelength causing a 120 phase shif
4. noise free measurements where a minimum signal to noise ratio of 20 dB in all seven octave bands is obtainable The standard specifically qualifies MLS and swept sine test signals for used with indirect measurement techniques but also states that Theoretically other mathematically deterministic pseudo noise random phase signals could be employed Period matched pseudorandom noise in Smaart fits that description and works well for STI There are a couple of potential advantages to using period matched noise over sweeps One is that it s important to conduct STI measurements at sound levels that reflect actual use of the system under test and the absolute level of sweep signals that start and stop can be ambiguous Another is the issue of distortion products potentially piling up at the end of the time record FFT based IR measurements made with logarithmic sweep signals Those need to be dealt with in some fashion either cropped off or windowed out lest they corrupt the measurement That isn t an issue with period matched noise although you still need to take care not to overdrive the SUT General caveats to using STI are that it is sensitive to strongly fluctuating background noise levels which can lead to overestimation of low intelligibility systems or underestimation of scores on the high end When measuring in the presence of fluctuating background noise at least three measurements should be taken and their results averaged to re
5. An acoustical impulse response with its common component parts labeled This is a semi log time domain chart with time in milliseconds on the x axis and magnitude in decibels on the y axis Propagation Delay The time that it takes for direct sound from the sound source to reach the measurement position is the propagation delay time This may include throughput delay for any DSP processors in the signal chain in addition to the time that it takes for sound to travel through the air 3 Page Copyright 2015 Rational Acoustics LLC All rights reserved Arrival of Direct Sound Since the shortest distance between two points is always the straightest line the first thing we expect to see when looking at an impulse response IR is the arrival of direct sound from whatever sound source we re using to stimulate the system under test Depending on what we re trying to learn the source could be an installed sound system an omnidirectional loudspeaker brought in specifically for measurement purposes a balloon pop or a shot from a blank pistol or in a pinch maybe hand claps or someone slamming a case lid shut In most cases we d also expect the first arrival to be the loudest and correspond to the highest peak we can see in the IR and in most cases we d be right There can be occasional circumstances where that might not turn out to be strictly true but in the vast majority of the cases it should Discrete Reflections After the arrival direc
6. also save it to a text file by clicking the Save button Frequency Domain Analysis Selecting Frequency as your graph type in the main graph area automatically Log e 6 Log transforms the IR into the frequency domain to show you its spectrum The S amp F Lin Frequency graph has frequency in Hertz on the horizontal x axis and 18 ETC y Frequency magnitude in decibels on vertical y axis The Smoothing control in the 2 Spectrograph upper left corner of the Frequency graph works exactly the same way as 36 Histogram smoothing on the real time transfer function display Smaart can calculate arbitrary length DFTs in IR mode to give you the spectrum of virtually any subset of the IR time record that you care to zoom in on Time and frequency domain displays are linked so that zooming in on a time domain graph Lin Log or ETC automatically changes the Frequency display to match When the entire time record is selected there s an assumption that you are analyzing a dual FFT IR measured in Smaart and so no data window is used in calculating the spectrum in that case Smaart automatically uses a tapered data window when transforming any subset of the time record so if you are analyzing an IR file from some other source or a file that has been cropped to less than its original length you may see better results if you zoom in slightly in the time domain Tapered data windows significantly attenuate data at the edg
7. an audience in place For convenience instead of measuring the full 60 dB decay time the decay slope in dB per second is measured over a 30 dB range preferred and then doubled to work out the RT60 The slope is conventionally measured between the 5dB and 35cB points as long as the lower point is at least 10dB above the noise floor If the noise floor is exceptionally high it may sometimes be necessary to measure the decay slope over a smaller level range In that case a 20 dB range may be measured instead and then tripled to normalize to equivalent 60 dB decay time According to ISO 3382 reverberation time measured over a 30 dB range is called 730 and To signifies a 20 dB measured range In their notation scheme the letter T by itself stands for 60 dB decay time and so hopefully it is understood that the 20 and 30 refer only to the measured range The stated reverberation time in seconds for either figure is the equivalent 60 dB decay at the measured rated of decay Typical values are as follows in practice of course acceptable ranges vary with expectations based on the venue size Talk studio less than 0 5 seconds Conference classroom up to 1 second Musical theatres 1 to 1 5 seconds Chamber music opera venues 1 5 to 2 seconds Symphony halls 1 5 to 2 5 seconds 55 Page Copyright 2015 Rational Acoustics LLC All rights reserved Appendix C Sound source characteristics A sound source s positioning orienta
8. are pointed Figure 4 shows a zoomed in view of the linear Lin scaled impulse response of a 2 order Butterworth bandpass filter with normal and inverse polarity Cutoff frequencies for the filter are 400 and 1600 Hz It s easy to see that the peaks in the two IRs are pointed in different directions relative to each other Unfortunately this doesn t necessarily tell you which one is correct But if you measured three like devices and one was different you might reasonably say that the majority rules Or if you measured two like devices and found opposite polarity and one of them sounded better it s possible you might have found the problem 16 Page Copyright 2015 Rational Acoustics LLC All rights reserved 21 47 ms 0 55 181 11 ms 0 04 159 64 ms 0 59 80 Lin F room wav 40 Qe S aoli 4 BO L 7 20 100 180 60 240 4270 500 5BO 660 740 820 900 980 20 60 100 140 180 220 260 300 340 380 Figure 5 Zooming in on a Linear Lin time domain view of room wav and using the cursor readout to find the relative arrival time of a prominent discrete reflection Energy Time Curve ETC The impulse response represents a 2 D graph of a 3 D event the magnitude and phase of the energy arrival over time With magnitude on the vertical y axis and time on the horizontal x axis phase ends up being represented on the z axis which is effectively lost in this view Consequently in the linear and log views of the IR energy arr
9. as we get into reverberation times and early to late energy ratios Smaart includes complete sets of octave and 1 3 octave bandpass filters for the octaves between 16 Hz and 16 kHz assuming 48k or higher sampling rate at lower sample rates you lose some of the upper bands Bandpass filtering in Smaart is done non destructively on demand To see a filtered version of the IR select which set of filters to use Octave or 1 3 Octave on the Filters selector then select the band that you want to look at from the Band list Smaart s bandpass filters have linear phase response and their magnitude response satisfies the most stringent Type 0 tolerances for octave and fractional octave bandpass filters specified in IEC 61260 and ANSI 1 11 If you would like to see the magnitude response of the bandpass filters you can load the wave file 1samplePulse wav and bring up the Frequency WM Octave x graph then step through the Bands list to see each filter Bandpass filtering MA f R Broadband applies to all main display types except the Histogram chart which is already 16 Hz filtered into bands It does not affect the small graph in the navigation pane 31 5 Hz Note that filtering the impulse response will clear the Spectrograph display if ate Hz present and require a recalculation by clicking the Calc button again 250 Hz 500 Hz Discrete Reflections 1 kHz 2 kHz Reflections are a complicated subject because humans are very go
10. bandpass filters in the previous section you were actually performing a direct IR measurement on the filters using an ideal impulse Unfortunately stimulus signals like that don t exist in the physical world When we need to measure the impulse response of an acoustical system directly we end up using stimulus sources that are less than ideal Blank pistols and balloon pops are common sources Signal cannon spark gaps fireworks and even spot welders have been used The problem with all of these is that their spectral content is not uniform their envelopes are not instantaneous they may not really be as omnidirectional as one might guess and all of these factors will vary to some extent from one measurement to the next This introduces uncertainly from the start as to which part of the completed measurement is stimulus and which is response It also limits the repeatability of test results For this reason systems such as Smaart that indirectly infer the response of a system to an ideal impulse have become more the tools of choice these days Indirect Dual Channel IR Measurement Indirect impulse response measurements are made using dual channel measurement techniques that mathematically estimate the response of an SUT using continuous or periodic test signals Three of the four indirect IR measurement methods we can name require specialized test signals such as sweeps Time Delay Spectrometry and Direct Convolution or specialized noise Hadama
11. channel file support or optimization for working with files more than a few seconds in length and of course a lot the IR analysis capabilities are irrelevant for other type of audio data 8 Page Copyright 2015 Rational Acoustics LLC All rights reserved 3 Cursor Readout When measurement data is present the cursor readout displays numeric coordinates for the cursor location s as you move your mouse over the graphs areas Numeric coordinates are provided here for the cursor location in units of time amplitude magnitude and frequency as applicable to graph type 17 23 ms 11 18 dB 59 23 ms 17 02 dB 42 00 ms 5 84 dB For time domain graphs Lin Log or ETC in the main graph area s there are three sets of coordinates as show above From left to right they are the location of the locked cursor which typically marks the highest peak in the impulse response the movable mouse cursor coordinates and in brackets on the right the difference between the locked and movable cursors Note that time coordinates can optionally be displayed as both time in milliseconds and equivalent distance traveled based on the currently specified speed of sound 4 Sound Level Meter The large numeric readout in the upper right corner of the Smaart v 7 Lett real time mode window can be configured to function as a standard Sound Pressure Level SPL meter an Equivalent Sound Level LEQ 4 d 8 E meter or a peak signal level meter calibrated to d
12. feet or 345 meters per second at typical Figure 27 The dk Mogg hfg size se ettor s owing the time constant room temperatures the speed of sound increases with temperature f in milliseconds PAPIT FFT size so if it is very hot where you are working you might adjust your K 42 MS ers estimate upward a little or downward if it s cold 1k 21 ms Band 512 7 10 ms For reverberation times one to two seconds should at least get you in 256 5 ms the ballpark for most theaters and auditoriums Stadiums and other 128 2 ms large structures can have much longer reverb times There s never any harm in measuring over too long a period so you may want to err to the high side If you make a preliminary measurement and you are happy with the results you might even be done If not you can adjust accordingly and measure again Note that as a rule lower frequencies tend to decay more slowly than highs meaning that the limiting factor may be the reverberation times in the lowest octaves that your stimulus source can excite So be sure to check the lower bands when estimating reverb times FFT Size Time window For dual channel measurements the duration of the measurement is determined by the FFT time constant that is the time required to record enough samples for a given FFT size at whatever sampling rate you are using In Impulse mode Smaart gives you the time constant in milliseconds along with the frame size in samples f
13. frequencies two per octave for a total of 14 STIPA is typically measured directly using a special test signal that excites all 14 frequencies at the same time so that the measurement can be completed in a single pass STIPA measurements can be completed in a few seconds and have been found to correlate very well with the more rigorous full STI STIPA is currently validated only for male speakers 34 Page Copyright 2015 Rational Acoustics LLC All rights reserved In Smaart of course we measure STI indirectly from the impulse response and the full STI measurement takes no longer to perform than a typical direct measurement of STIPA Smaart does provide figures for both STI and STIPA however STIPA in our case is more properly termed STIPA IR since it is based on IR data rather than measured directly It is provided for informational purposes e g to facilitate comparison with readings from hand held STIPA meters and is literally just a subset of the full STI measurement calculated from exactly the same measurement data ALCons ALCons sometimes called ALCons because it is stated as a percentage stands for Articulation Loss of Consonants Consonants are critical to speech intelligibility as they are short in duration and tend to get lost more easily than vowel sounds that are voiced over a longer period of time ALCons was originally conceived as an estimate based on distance of the listener from a sound source the directivity Q o
14. have to measure over a longer period than you would with a purpose built signal to get comparable results It is left up to the operator to decide how much averaging or how long a time window to use and the actual dynamic range of the SUT is ambiguous This can be a critical factor in speech intelligibility measurements Reducing Noise in IR Measurements Made Using Random Stimulus Signals There are three basic things you can do to improve the dynamic range of measurements made using random test signals The first is to delay the reference signal to match the timing of the measurement signal so that the data windows line up You should always do this when measuring with random signals The second is to evaluate the system over a longer period of time by increasing the DFT size or by averaging multiple measurements or both The third is to simply measure louder which also applies to deterministic and direct IR measurements in that case you re increasing the signal to noise ratio of the measurement by increasing the level of the actual signal rather than statistically 41 Page Copyright 2015 Rational Acoustics LLC All rights reserved Figure 24 The effect of averaging on an IR measurement made using random stimulus signals In theory each doubling of the number of averages increases signal to noise ratio by 3 dB Averaging works by inducing regression to the mean in random components of the IR that is to say the noisy part Let s
15. lower room absorption preferably with plenty of diffusion and a good balance between early and late decay times will add air to our sound and prevent it from becoming too dry Directivity index DI An alternative directivity figure that indicates the concentration of radiated power in dB is the Directivity Index DI Its value in dB is DI 10 x logQ So assuming the listener is on axis if Q 2 DI 3dB if Q 4 DI 6cB etc Q and DI figures in practice Large radiating surfaces tend to be more directional than small ones Whilst good far field summation occurs perpendicular on axis to large sources less than perfect summation occurs off axis due to phase variations between waves emanating from different areas of the surface This can cause minor off axis lobes at some frequencies and cancellations at others Cod Kot NK ia EUR REY Jee MENA PREC ASA DIAMETER 1 DIAMETER 1 2 DIAMETER X MAJOR LOBE MINOR LAJ LEI LOBES Sha Std sO LX SMO CK E KAD A LX Greg d Ee Q TANS ei oe OS EK DIAMETER 2 DIAMETER AA DIAMETER GA 0 40 Q 160 Q 316 DI 16 dB Dl 22 dB DI 25 dB Courtesy JBL Professional Harman 57 Page Copyright 2015 Rational Acoustics LLC All rights reserved The above illustration shows the polar patterns plus the relevant directivity factor Q and directivity index DI of baffle mounted piston like sound sources e g stiff loudspeaker cones for various diame
16. measurement mode control Clicking the start gt button without the record button punched in kicks off a dual channel IR measurement With the record button e activated Smaart becomes a single channel digital recorder and records just the measurement signal channel of the selected signal pair The idea is that you would start recording and pop a balloon or fire your starter pistol or whatever then click the stop button to end recording and display your single channel impulse response measurement 10 Page Copyright 2015 Rational Acoustics LLC All rights reserved The Continuous Cont button causes the dual channel measurement routine to run continuously starting over again automatically each time it finishes a measurement until you tell it to stop by clicking the stop button The results of the last measurement are displayed while it s recording and processing the next measurement Click the stop button to end the measurement Underneath the recording controls there s a status bar to let you know what Smaart is doing when a measurement is in progress since IR measurements can take a while to complete Below that are input level meters and reference signal delay controls for the selected signal pair The button sets the delay time for the measurement to the highest peak in the impulse response when clicked If you are measuring using a random stimulus signal and the delay is not already set you would run the measurement onc
17. on the x axis and magnitude in decibels on the y axis The combination of locked and movable cursors enables you to find time and level differences between any two points on the plot Time coordinates can optionally be plotted with equivalent distances as shown above The pair of coordinates on the far left in the cursor readout at the top of the frame is the locked cursor position which is set to the highest peak in the IR The middle pair of coordinates in green is the absolute location for the movable cursor and the rightmost pair in brackets is the difference between the first two Linear Time Domain Display A linear Lin time domain chart plots the same data as the Log IR but on a normalized linear amplitude scale where amplitude values are given as a percentage of digital full scale This view tends to be of limited usefulness for acoustical analysis in general however it can be a very good tool for finding discrete reflections particularly when measuring in an empty hall before an audience arrives In this case using the linear IR view can help you to identify hard reflections that might be masked by the diffuse reverberant field on a logarithmic display only to become much more obvious and audible often on stage to the consternation of opera singers once there is an audience in place and the reverberant levels decrease Another thing the Linear IR can tell you that the Log and ETC graphs can t is relative polarity For example you co
18. parts contain much more information with regard to measurement procedures and statement of results than we have presented here For anyone interest in getting into the nuts and bolts of techniques used for indirect IR measurement ISO 18233 Acoustics Application of new measurement methods in building and room acoustics is a good place to start If you are interested in making speech intelligibility measurements using STI or STIPA IEC 60268 16 Sound system equipment Part 16 Objective rating of speech intelligibility by speech transmission index Edition 4 2011 or later is really a must read Further Reading Two books that we found particularly useful in preparing this document were Ballou Glenn ed Handbook for Sound Engineers Third Edition Focal Press Toole Floyd Sound Reproduction Loudspeakers and Rooms Focal Press Classic Reference Works for Acoustics Beranek Leo J Music Acoustics and Architecture Wiley Olson Harry F Modern Sound Reproduction Van Nostrand Reinhold Rigden John Physics and the Sound of Music Wiley Jones D S Acoustics and Electromagnetic Waves Clarendon Press Kleppe J A Engineering Applications of Acoustics Artech House Rossi Mario Acoustics and Electro acoustics Artech House Berg Richard E and Stork David G The Physics of Sound 2nd Ed Prentice Hall Borden G J and Harris K S Soeech Science Primer Physiology Acoustics and Perception of Speech 2nd Ed Willi
19. point is always 5 dB down on the reverse integration curve from the point corresponding to the arrival of direct sound The end point of the range is 30 dB down the curve from the starting point provided that it is at least 10 dB above the noise floor if not a 20 dB range may be used In either case the measured decay time is extrapolated to the equivalent 60 dB decay time In ISO 3382 parlance these are referred to as T20 or T30 Early decay time EDT is conventionally measured from the arrival of direct sound down to 10 dB below it on the integration curve Like reverberation time EDT is also normalized to 60 dB decay time 21 Page Copyright 2015 Rational Acoustics LLC All rights reserved D A 9 56 dB EDT 0 753 sec AT60 1 067 sec 56 3 dB sec Reverberant Decay 42 JI s T Slope Lr1 Lr2 Ld Le 125 5 ms 10 00 dB AR Ld Ln 694 3 ms 64 51 dB Lri Lr2 533 4 ms 30 00 dB 54 60 i E 56 4 Reverse Time Integration i Saddle Point Wa GER M a gll i a 72 f a 50 150 250 350 450 550 650 7950 850 950 1 05k 1 15k Figure 9 A log IR display with all the bells and whistles The IR is the 500 Hz octave band of theater wav Clicking the Schroeder and RT60 buttons displays the reverse time integration curve and the start and end points for the EDT and RT60 evaluation ranges on Log or ETC charts The positions of all the level markers Ld Le Lr1 Lr2 and Ln are user adju
20. regarded as an important metric in the acoustics of musical performance spaces and also classrooms auditoriums and cinemas where it is used as a rough predictor of speech intelligibility Reverse Time Integration Reverberation time is calculated from the reverse integration of an impulse response that has been filtered into octave bands Conventionally the 125 Hz to 4 kHz bands are evaluated Reverse time integration is also called Schroeder integration after Dr Manfred Schroeder whose brain child it was It is a simple thing in concept but it can be a little tricky to do well In theory you just start at the end of the time record and work your way back to the beginning tallying up the squares of each sample in the IR as you go Acommon problem however is that the integration will flatten out when the reverberant decay slope runs into the noise floor of the IR This can lead to overestimation of the reverberation time particularly if the IR has limited dynamic range and or a lengthy noise tail 20 Page Copyright 2015 Rational Acoustics LLC All rights reserved The most straightforward solution for this problem is to find the point in the IR where the decay slope meets the noise floor sometimes referred to as the saddle point and begin the integration there rather than at the end of the file The location of the saddle point in an IR is notoriously difficult to estimate automatically though Smaart 7 5 or higher uses a propri
21. reverberation times in the upper octaves ISO 3382 1 unequivocally states that the sound source shall be as close to omnidirectional as possible and provides criteria for assessing the omnidirectionality of a prospective source ISO 3382 2 specifies measurement procedures for three levels of accuracy in reverberation time measurements Survey quick and dirty Engineering pretty good and Precision very good For the Precision method the requirements for the excitation source are identical to those specified in 3382 1 but 3382 2 goes on to say that For the survey and engineering measurements there are no specific requirements for the directivity Clearly the spirit of the law is that omnidirectional sources are preferred for reverberation time measurement but the standard does leave a little wiggle room when measuring in ordinary rooms as opposed to formal musical performance spaces 44 Page Copyright 2015 Rational Acoustics LLC All rights reserved Differences between reverberation times measured using directional versus omnidirectional speakers will tend to be most significant in the frequency ranges covered by the most directional loudspeaker elements As such these are likely to be most problematic in the 1 kHz though 4 kHz octave bands 4 kHz being the highest octave band normally evaluated for reverberation time An informal survey of IR measurements we happened to have on hand that were made using directional PA s
22. say you take a signal any signal maybe an impulse response and mix it with random noise Obviously you get a noisy signal There is no way to tell just by looking which part was signal and which was noise But if you take several copies of the same signal and mix each one with different noise then average all of them together the noise component of each noisy signal being random and different in each case should start to average toward zero the theoretical arithmetic mean for random audio noise while the signal parts being the same in every case should average out to themselves Of course all of this depends on the assumption that the signal part of the signal is the same in every case When working indoors that should generally be a safe assumption After all we are working with what we assume to be linear time invariant systems in a fairly controlled environment where the worst that could probably happen from one pass to the next is a blast of hot or cold air from an HVAC system causing a slight change to the speed of sound It might be a larger concern if you needed to make an IR measurement outdoors under windy conditions for some reason In any scenario where there might be a possibility of any significant time variance during the measurement period you would probably be better off increasing the measurement time window and or using a period matched stimulus signal rather than upping the number of averages In theory aver
23. the average normal standing or seated height of human ears in the space 5 6 for standing and AO for seated adults 3 6 standing and 2 6 for seated children The microphone shall be no closer than AO from any sound reflecting surface or AO from the intersection of two intersecting reflecting surfaces or 8 O from the intersection of three intersecting reflecting surfaces SMPTE 202M recommends that microphones be placed In indoor theaters at position S and position R should it exist and at a sufficient number of other positions to reduce the standard deviation of measured position to position response to less than 3 dB which will typically be achieved with four positions It is recommended that measurements be made at a normal seated ear height between 1 0 m and 1 2 m 3 3 ft and 4 0 ft but not closer than 150 mm 6 in from the top of a seat and not closer than 1 5 m 4 9 ft to any wall and 5 0 m 16 4 ft from the loudspeaker s Position S generally works out to be a little left or right of the approximate center of the room on the main floor Position R is for balconies We can see that these are all in general agreement depending to some extent on frequency even though one talking about reverberation time another is for background noise surveys and the third is for RTA measurements of cinema sound system They are probably also not out of line with positions you would intuit
24. to 0 each FFT is calculated from unique data At 50 overlap the darker shaded areas are shared by successive constant in every case At 512 points time resolution FFTs Our FFTs are drawn as flattened half the FFT time constant is a respectable 10 7 milliseconds circles to suggest a tapered FFT data window notice how its energy is spread across the full FFT time but the FFT frequency bins are spaced almost 100 Hz apart Increasing the FFT size to 4K points gets you 12 Hz frequency resolution but smears the peak in the IR over an 85 ms time range The other factor affecting the time resolution of the spectrograph is the Overlap percentage that you specify in the upper right corner of the graph When Overlap is set to zero as in Figure 15 each successive FFT slice of the Spectrograph is calculated from unique time domain data each frame beginning where the last one ended When Overlap is set to any non zero value each successive FFT frame shares some percentage of its data in common with the previous frame s The FFT time constant is still the FFT time constant but more overlap can sometimes allude to if not exactly restore some missing detail on the time axis as FFT size is increased in addition to producing smoother blending between slices See Figure 16 H TE Lin i 6dbOctimpulse wav in Lin 6dbOctimpulse wav jas Lin 6dbOctimpulse wav 0 gegen Ar P a ne 0 ee gegen anil ee ame 0 re
25. to utter their next That was actually the problem being investigated in the IR measurement shown in Figure 18 where a high level reflection was arriving at about 160 ms which is close to the average syllabic rate for normal conversational speech Low order early reflections may be visible on time domain plots as individual peaks following the arrival of direct sound Later arrivals can show up as spikes protruding from the reverberant decay slope On the Spectrograph plot higher level broadband reflections can often be identified as distinct vertical streaks when you run the dynamic range controls up and down particularly the Max setting They tend to be most problematic when arriving at longer delay times and relatively high levels compared to the level of the diffuse reverberant field A pretty good rule of thumb is that the later the arrival the lower in level it needs to be in order to be perceived as beneficial or neutral Another is that our tolerances for reflected sounds and reverberance tend to be wider for music than speech In general Smaart is very useful for identifying problematic reflections however your ears are probably still the best tool for evaluating their relative significance or severity Reverberation Time Reverberation time commonly referred to as T60 or RT60 or somewhat less commonly as T30 T20 or simply T is the time required for reverberant sound energy in a space to decay by 60 dB from an excited level It is
26. use to filter the IR on the fly to see how reverberant decay and other characteristics change with frequency The combination of locked and free cursors on time domain displays enables you to find the relative arrival time and amplitude differences between any two points on the plot The difference between the two is shown in the cursor readout If the Milliseconds and Distance option is selected in the Cursor Time Readout section of the General options page Options menu gt General Smaart will also give you equivalent distances for time coordinates based on the current Speed of Sound settings To move the locked cursor to an arbitrary point on the plot hold down the Ctrl key Cmd key on Mac on your keyboard while clicking with your mouse on a point that you want to mark Pressing Ctrl Cmd P resets the locked cursor to the highest peak IR 14 Page Copyright 2015 Rational Acoustics LLC All rights reserved 17 23 ms 19 4 ft 1118dB 171 67ms 193 5ft 35 19dB 154 44ms 174 1 ft 24 01 dB 70 Lin theater wav 20 100 180 260 340 420 500 580 660 740 820 900 980 1 06k 1 14k 12 18 24 30 36 42 48 E L E si PUTT lt Al ou BRSBIERERORRRR Nd ME AN ee een wend um L i WM ui Se COO NI WESSEN SSRRRSRSERERIDEIRESRSESIRRIRERIEIESRGRES RIES RSR eee 20 100 180 260 340 420 500 580 660 740 820 93900 980 1 06k 1 14k bi BIRA ngaa EERE Figure 3 The logarithmic Log time domain graph plots time
27. used the weighting tables for Articulation Index and STI and other scales of their own devising with similar results For music he used a simple average of the 500 Hz 1 kHz and 2 kHz octave bands We know of no applicable standards for this metric and it has been suggested extending the frequency ranges Marshall used an octave higher for speech and two octaves higher for music that it might be useful but hopefully this example provides a useful starting point for evaluation The Histogram Display Selecting Histogram as your display type for an IR mode graph plots a chart of all reverberation times or early to late energy ratios by octave or 1 3 octave bands The type of data the Histogram displays and the resolution are selected by means of the list control in the upper right of the graph You can change the Histogram to a line chart by opening the Impulse Response options page Options Menu gt Impulse Response and selecting Plot as Line under Histogram Settings Smaart v7 ee File Options Mode View Zoom Cursor Help Left 80 Lin theater wav 4 2 8 40 be of Basel Sow Max 82 5 O pad DL CAI Bands T60 Real Time 50 250 450 650 850 1 05k 1 25k 1 45k 1 65k 1 85k 205k 2 2 S EH Schroeder Histogram Oct RT60 V PinkNoise 15dB Leit All Bands x EDT D R cio c35 c50 cso 1 09 9 54 4 52 0 67 3 13 4 76 1 13 9 55 2 32 0 25 0 61 4 76 0 84 9 55 2 98 1 36 0 08 4 38 0 83 9 55 0 86 0 52 1 09 3 56 0 83 9
28. 5 so the usual 6 dB doubling characteristic is modified by the extra element coming into play This phenomenon is similar to the near field effect mentioned earlier which occurs close to any large sound source As Victor continues his journey further away he then benefits from the vector sum of more and more elements and these partially compensate for the usual 6 dB doubling of distance each element in 59 Page Copyright 2015 Rational Acoustics LLC All rights reserved isolation would have Victor will benefit from this low 3 4 dB doubling of distance attenuation region all the time there are extra elements available However Victor will eventually run out of these extra elements and the radial attenuation characteristic will revert back to the usual 6dB doubling of distance This point in Victor s journey is called the transition distance see next illustration Note that the transition distance for a true line source is proportional to frequency and to the line length S l Line length x frequency Transition Distance E 2 x speed of sound Allowing for the fact that the speed of sound is temperature dependent transition distances for a typical 20 ft 6m straight long line source at 68 F 20 C will be approximately This suggests that high frequency propagation would be very efficient But in practice HF energy will be significantly reduced by air absorption by up to 0 1dB meter worse case and of cours
29. 55 2 47 3 30 4 59 6 32 0 84 9 54 1 97 3 31 4 66 6 12 0 67 9 54 2 60 3 89 5 79 7 91 0 25 9 57 5 04 6 25 10 24 12 14 0 25 9 89 7 54 8 81 13 58 15 99 CIS 0 84 Bass Ratio 1 11 Alcons S 4 12 STI 0 69 T Low 1 17 Alcons L 5 25 63 125 250 500 1k 2k 4k 8k 16k STIPA 0 70 T Mid 1 05 Detail Figure 11 The Histogram graph and All Bands Table The All Bands Table in Smaart collects reverberation times and early to late energy ratios for all octave and 1 3 octave bands in a single table Speech intelligibility metrics STI and ALCons are displayed here as well The Histogram chart can plot any column of the All Bands Table as a bar graph or line chart in octave or 1 3 octave resolution 25 Page Copyright 2015 Rational Acoustics LLC All rights reserved The All Bands Table Clicking the All Bands button in Impulse mode in Smaart 7 brings up a report window containing just about every acoustical quantity that Smaart can calculate automatically from an IR for each octave band and 1 3 octave band where applicable see Figure 11 Speech intelligibility metrics STI and ALCons are displayed here along with Bass Ratio T Mid and T Low which are calculated from reverberation times for the 125 Hz to 1 kHz octave bands Clicking the Copy button in this window copies the entire table to the operating system s clipboard in tab delimited ASCII format suitable for pasting into a spreadsheet or any other program that accepts ASCII text You can
30. Smaart 7 Impulse Response Measurement and Analysis Guide 20 36 42 48 54 Ka 66 72 78 84 90 96 sgg 8888 rege rational acoustics Copyright 2015 Rational Acoustics LLC All rights reserved MOGUC ON EE 1 SCODE dd be ld ele GT e eet EEN 1 de MEET reponse Eege Ee 2 Anatomy of an acoustical impulse response cccceseccceesececeesececceeecceceusececeenecesseneceeeensecessunecessenecessenes 3 FAR oye xe ne RR 3 EENEG 4 Discrete d e den EEN 4 Early Decay Reverberant Build up and Reverberant Decay cccceecccessceseseceeseeneseenseeenseneeseeeeeeees 4 Noise tele EE 4 Uses for impulse response measurement data 5 Delay Time TEE EE 5 RETICCEIO MAL ELE 5 Reverberation time 160 RIGO x2 ee eege eher eet 5 Early Decay TIMER AR EE 5 Bam lO ate Cnerey TIOS ioin N E N E O EN 6 Speech Intellieibility Modeling cernens a a a saa ieeet 6 2 A Quick Tour of the Smaart 7 Impulse Response Mode User Interface 7 E ein We EE 7 2a Main Ee ee Re E 8 5 CUNSOR e e 9 4 Sound Level Meter totic cscezces arate Sere vaccinate oattocets ated ues vs nescence eevee ecto as aatescure oman eae aaedonne eeeaee 9 Di Data IS Olay CONUS eaa eatin sss ET rece ee ee erie a seen a sane ee cee 9 6 Sienal Generalo EE 10 ker Fi SIZEANGAVELACINEG e nd EE 10 S OPU SOURCE EEN onn a T Ee A 10 9 Live Measurement Controls orense S 10 NOG Bandpass GIE 11 Additional Options for IR Meaesurement 11 General Options Pertai
31. The Histogram chart in IR mode plots the values found for any column in the All Bands table band by band for all octave or 1 3 octave Selector controls are found in the upper right corner of the chart in the main window By default Smaart plots the histogram chart as a bar graph Selecting Plot as Line in Histogram settings causes this chart to be plotted as a line graph instead 13 Page Copyright 2015 Rational Acoustics LLC All rights reserved 3 Analyzing Impulse Response Data Smaart provides a powerful set of tools for analyzing impulse response data in both the time and frequency domains Time domain analysis tools include logarithmic and linear time domain views Energy Time Curves octave and 1 3 octave bandpass filters reverse time integration and automatic calculation of common acoustical parameters such as EDT RTeo and clarity factors Frequency domain analysis tools include spectrum analysis of arbitrary time ranges and the Spectrograph If we were discussing Smaart s real time measurement and analysis mode we would almost have to pause at this point to set up and start actively measuring some kind of sound source in order to have something to analyze But in IR mode measuring and analyzing are generally two separate things that we can talk about separately Data analysis in IR mode is an off line post process affair that works the same whether we re onsite actively measuring a system or working with an impulse response recorde
32. aging two IR measurements or doubling the FFT size used for a single IR measurement should improve signal to noise ratio of the measurement by 3 dB Note that both result in doubling the measurement time which is really the key to the whole thing Each additional doubling 2 4 8 16 of the measurement duration should theoretically get you another 3 dB although in practice you might reach a point of diminishing returns at some point Selecting Excitation Sources and Positions Excitation source positions should be places that sound would normally emanate from when the system under test is in service If the loudspeakers you are using to excite the room are the places that sound normally comes from then you ve got that part covered Otherwise an omnidirectional sound source of 42 Page Copyright 2015 Rational Acoustics LLC All rights reserved some kind should be placed on the stage podium lectern pulpit or whatever location s that would best simulate normal use of the room system and at an appropriate height Minimum Distance from Sound Sources Another general requirement for measurement positions used for room IR measurements is that they need to be located far enough away from loudspeakers or other sound sources being used to excite the room to ensure that the measurement is not unduly dominated by direct sound ISO 3382 2 provides the formula shown in Eq 1 for calculating the minimum distance dmin for any measurement position fr
33. aluating the relationship between beneficial direct sound and early reflections that a listener hears versus the amount of potentially detrimental reverberation and noise than inferences made from the early and reverberant decay rates Speech Intelligibility Modeling Early to late energy ratios such as C35 and C50 have long been used as objectively measurable predictors of subjective speech intelligibility In the 1970s Victor Peutz came up with Articulation Loss of Consonants ALCons a predictive metric for intelligibility of intelligibility based in the volume of a room and its reverberation time the directivity of loudspeakers and distance from source to the listener Later on Peutz revised the equation to use a direct to reverberant energy ratio in place of volume distance and loudspeaker Q making ALCons a directly measureable quantity More recently the speech transmission indexes STI and STIPA have emerged as metrics that are generally more robust All of these can be calculated from the impulse response of a system 6 Page Copyright 2015 Rational Acoustics LLC All rights reserved 2 A Quick Tour of the Smaart 7 Impulse Response Mode User Interface If you already know your way around IR mode in Smaart 7 you can probably skip this section but it might not hurt to at least skim over it If you are new to IR mode then introductions are in order To get to IR mode in Smaart select R Mode from the Mode menu press the I key o
34. ams and Wilkins 52 Page Copyright 2015 Rational Acoustics LLC All rights reserved Appendix B Room Volume Absorption and Reverberation time RT60 Reverberation time is usually measured or calculated for a 60dB decay range RT60 The following information will help you to appreciate the factors that affect reverberation time so that after measurements have been made appropriate changes can be suggested where necessary RT60 is heavily dependent on the room volume the amount of absorption in the room and air absorption at higher frequencies A popular approximation based on Sabine s formula is 55 3V a d 4mV where RT is the time taken for the reverberant sound energy to decay by 60dB in seconds V is the volume of the room in m3 c is the speed of sound in m s varies with temperature A isthe total sound absorption of room materials in m Sabines i e the sum of surface areas each multiplied by its respective sound absorption coefficient m is the intensity attenuation coefficient of air per meter varies with temperature and humidity Material absorption coefficients Typical absorption coefficients Material Typical mid band sound absorption coefficient Marble 0 01 Plastered walls 0 02 Bare brick 0 03 10mm Plywood 0 09 10mm mineral wool 0 60 25mm polyurethane foam 0 70 Acoustic ceiling tile 0 72 Audience member on upholstered seat 0 88 Note that absorption coefficients vary with frequency
35. ccurate RT analysis The crossover between the Schroeder region and the Modal region is known as the Schroeder frequency This frequency only marks a cross over region where modes start to overlap by a fairly arbitrary factor of three It is not an exact science A popular approximation based on Schroeder s work is f 2000VRT As before RT is the time taken for the reverberant sound energy to decay by 60dB in seconds and V is the volume of the room in cubic meters For room volume in cubic feet the equation can be written as f 11885VPT y Practical measurement considerations Most of the above equations are approximations assuming an omnidirectional sound source homogenous sound distribution and the possession of accurate venue information In reality accurate venue information may be difficult to acquire and operating conditions may vary from event to event 54 Page Copyright 2015 Rational Acoustics LLC All rights reserved Reverberation characteristics are also likely to vary slightly across the audience area and early decay characteristics see later will vary significantly from seat to seat so accurate measurements over a practical range of operating conditions are recommended As most rooms have significant background noise levels the noise floor a full 60dB decay range isn t always available especially when the direct sound is at normal speech levels or as is possible with Smaart we measure a system with
36. cifications inaccurate venue drawings and a lively room s sensitivity to relatively small changes in array shape and tilt Prediction software can get us close to our design goal and will certainly enable us to work out requirements and budgets etc but we need Smaart to check that installed systems are meeting spec Smaart s ability to complete as built measurements even during shows with an audience in place if required makes it a very powerful verification tool Line arrays It should be noted that the term line array has expanded in popular usage in recent years to become a term for a relatively long vertical array of loudspeaker typically just one cabinet wide Manufacturers often imply that a line array system will give the user a text book line source radiation characteristic and many users assume they ll benefit from a radial attenuation of 3dB per doubling of distance rather than the 6dB doubling of a conventional array In reality there is rarely the budget for a straight line source column as it would have to stretch from the floor to the highest seat Most line arrays are much shorter for budget and safety reasons so have to be progressively curved from top to bottom in order to cover the seating areas from the furthest highest seat down to the front floor areas beyond the front fill coverage 58 Page Copyright 2015 Rational Acoustics LLC All rights reserved However the upper straighter par
37. d in a wav or aiff file Since we just talked about the IR mode user interface in the previous section let s dive right into actually using it while that discussion is still fresh in our minds Most of the examples in this section were created using a handful of wav files that have been included with this document Wherever applicable we will tell you which file was used and how to duplicate our settings so that you can gain a little hands on experience as we go if you like Our first example uses theater wav an IR measurement of a 400 seat historical vaudeville theater The measurement was taken from about the main floor seating area using a small horn loaded PA speaker positioned on the stage lip as the excitation source If you would like to load the file yourself open up Smaart switch to IR mode then select Load Impulse Response from the File menu and navigate to wherever the file resides on your hard disk to open it Time Domain Analysis Logarithmic Time Domain Display The time domain IR display with logarithmic Log amplitude scaling is probably the most familiar to anyone much accustomed to looking at acoustical impulse responses In this view you can find the arrival times of direct sound and early reflections and overlay the reverse time integration of the IR along with interactive widgets to calculate EDT and reverberation time on Log and ETC displays only Smaart provides octave and 1 3 octave bandpass filters that you can
38. did But at that point it would much more readable if you just dispensed with the bar graph and plotted a 2 D chart instead with frequency on one axis time on the other and magnitude indicated by color Figure 13d That s a spectrograph Generally the domain of a graph is the independent variable e g time or frequency which is normally assigned to the horizontal x axis but the spectrograph display has two independent variables You can orient it whichever way is most convenient In real time mode in Smaart we want to relate the spectrograph to other frequency domain graphs so we plot it with frequency on the x axis and time on the y axis In IR mode we most often want to look at it in the context of other time domain graphs so we do it the other way around Time goes on the x axis and frequency on the y axis 28 Page Copyright 2015 Rational Acoustics LLC All rights reserved To bring up the spectrograph in IR mode click on the graph selector Ae Log in the upper left corner of a main graph area pane and select Spectrograph Initially you are presented with a blank chart area j until you click the Calc calculate button in the upper right corner of op Frequency the graph pane The Spectrograph can eat up a lot of graphics 30 Spectrograph resources when it repaints and so we try to paint it only when 36 necessary Changing the time range selection in the navigation pane does not affect the spectrograph as it d
39. duce measurement uncertainty Also if the soeech source and some prominent source of interfering background noise are widely separated STI may underestimate intelligibility human hearing can be smarter than machines about that kind of thing STI is also sensitive to clipping or amplitude compression in the transmission channel but in our case those would also violate the linear time invariant system rule for transfer function measurements So don t do that Qualitative Thresholds In terms of specific levels to look for anything better than 0 80 STI is considered excellent STI scores between 0 61 and 0 80 are Good 0 45 0 60 is Fair 0 35 to 0 44 is Poor and anything less than 0 35 is atrocious Note that there are male and female versions of STI in the more recent versions of the standard Any time an STI number is stated without specifying whether it s for a male or female speaker the assumption is that it s the male version STIPA One of the problems with direct measurement of STI is that it takes a lot of time to make a measurement The modulation frequencies are so closely spaced that each one had to be measured separately There are 98 modulation frequencies in all 14 x 7 and the full direct measurement takes about 15 minutes to perform as a result STI for Public Address systems STIPA was developed to get around this problem STIPA is essentially the same measurement as STI but uses a subset of its modulation
40. e imperfect coupling at high frequencies due to rigging tolerances etc It is particularly important to remember that transition distance is proportional to the square of the line length if spectral balance is to be maintained over a wide area The length of the straight line section at the top of most arrays needs to be at least 6 of the maximum distance to be covered for spectrally balanced vocals Q What has all this got to do with measuring room acoustics using Smaart s Impulse mode A Smaart s ability to measure room phenomena live with an audience in place means that you will often be asked to assess acoustics under practical amplified show conditions using an installed PA system Many of these installed systems will be line arrays whether appropriate or not and a basic understanding of them will allow you to make intelligent recommendations 60 Page Copyright 2015 Rational Acoustics LLC All rights reserved Appendix D Boundary effects Acoustically small sources Acoustically small sound sources are sources whose dimensions are small compared with the sound wavelengths they produce Away from boundaries acoustically small sources tend to be omnidirectional A single driver 1m cube subwoofer would be a good example of an acoustically small sound source as its size would be less than a quarter of the wavelength at low bass frequencies At 80Hz for instance the wavelength would be just over 14 ft 4m
41. e paging system to estimate its intelligibility Selection of Measurement Positions The first and most obvious rule for selecting measurement positions is that you generally want to measure from places where one would expect to find listeners when the system under test is in service If a tree falls in the forest and no one is there to hear it who really cares if it makes a sound You might also want to give special attention to any areas where you think there could be problems Other than that it is kind of like taking an opinion poll If we measure from a single position we have one opinion of what the room sounds like If we sample from a several different locations we might reasonably expect to see some consensus emerge as to the most common characteristics of the system response and for position dependent differences to begin to average out The more measurement positions the lower the theoretical margin of error assuming the positions are chosen so as to be Statistically valid For the Survey method in ISO 3382 2 a single stimulus source location is measured from at least two measurement locations providing a theoretical margin of error of 10 for octave bands The Engineering method calls for at least two stimulus source positions and six independent source microphone combinations for a nominal accuracy 5 for octave bands or 10 in 1 3 octave bands The precision method calls for 12 independent source microphone combinations
42. e time that it takes for reverberant in a room sound to decay by 60 decibels from an excited state after the excitation signals stops It is one of the most widely used and in some cases perhaps misused quantities in room acoustics Although it is quite possible for two rooms with identical reverberation times to sound very different when evaluated band by band it can still give you some idea as to the overall character of the reverberant field in a given room In concert halls it can give you an idea of perceived warmth and spaciousness for music In auditoriums it is often used as a rough predictor of speech intelligibility Early Decay Time EDT Early decay time ends up being the decay time for direct sound and earliest lowest order reflections Since the earliest reflections tend to be the most beneficial in terms of separating sounds we want to hear from reverberation and background noise EDT can give you some clues about overall clarity and intelligibility in a room and or system EDT like RT60 is conventionally normalized to the time it would take for the system to decay 60 dB at the measured rate of decay 5 Page Copyright 2015 Rational Acoustics LLC All rights reserved Early to late energy ratios Early to late energy ratios are a direct measure of the sound energy arriving within some specified interval following the arrival of direct sound vs the energy in the remaining part of the IR These provide a more direct method of ev
43. e to get the delay time then click the button then run the measurement again You generally don t have to worry much about the delay time with deterministic IR measurements They work fine with the delay set to zero To zero out the delay time type a zero in the delay time field and hit the enter key 10 Bandpass Filters The broadband impulse response is useful for finding delay times but for most acoustical analysis purposes the IR needs to be filtered into octave ee Octave si bands or occasionally 1 3 octave bands Smaart includes complete sets Band 500 Hz M of octave and 1 3 octave bandpass filters for impulse response analysis Bandpass filtering is non destructive and is done on the fly whenever you need it All you have to do is select the filter set that you want to use Octave or 1 3 Octave using the Filters selector in the lower right corner of the IR mode window and then choose the center frequency for the band that you want to analyze from the Band list Options x Additional Options for IR General Spectrum Transfer Function Impulse Response Measurement Enabled C Some additional options pertaining to IR Port 26000 Connected Line Thickness measurement that don t appear on the main Foreground Trace 2 screen can be found in the General and Impulse Background Traces 1 Response options pages accessible from the Cursor Frequency Readout Options menu G Frequency O Frequency amp Wavelen
44. ecseeeseeeeseeeeeceseueseseueseneneees 37 Dual Channel IR measurement Using Period Matched Signals ccccccssecccsseecceseeceeeceseescensneeeeees 38 Dual Channel IR measurement Using Random Stimulus Signals cccssecccsseeeceseeceesceeeeceeeneeeeees 41 Selecting Excitation Sources and EES TEE eege EE 42 Minimum Distance from Sound Sources ressens a A R 43 ii Page Copyright 2015 Rational Acoustics LLC All rights reserved Directional Loudspeakers and Reverberation Time 44 Selection Of Measurement Bosttions ceria AE A E A EAA 45 selecting Measurement Parameters eier Eege 46 MPU SOUC E 47 Excitation te EE 47 MOULLEVE EE 47 Vue Ee Tee E 48 PNET AGING and OVELIAD VE 48 Pushing the Button and Making the Measurement sssssesesssenssresererssrresrersrrresreeseersrrroseerseersereesrersee 49 SAVIN VOU VY OU EEN 49 Recap Recommended Default Settings for Dual channel IR measurements sesssrererrrerrrrerrrerrreen 50 SIE ll TYPE aaran AS oti ay teladabee N N N a eeaee 50 FET ECHTER 50 Excitation ME 50 HPU ECVE E 50 Delay ME canser a a EEN canes eenantn Uae aareui ys hale donee uisies 50 Appendix A Applicable Standards and Further Reading ccscccsssccccsscecenececenececenceeeenceesencesenecesseeeeees 52 Farther REJON EE 52 Classic Reference Works Tor ACOUSUCS crosier E a A E N 52 Appendix B Room Volume Absorption and Reverberation time RT60 esssessssssseesessrsesrresesreresrrereseeress 53 Mat
45. el at the listening position If we use directivity factor Q to quantify a sound source s directional properties then for on axis listeners the directivity dependent relationship between source sound power level and the total direct reverberation listener position sound pressure level approximates to Ly Ly logio K T p The directivity factor Q can also be used to estimate the critical distance Dc Dc 0 14 x QR For the previous two equations Lp is the total direct reverberation sound pressure level at the listening position Ly is the source s sound power level in dB ref to 10 Watts isthe directivity factor 1 for spherical 2 for hemispherical 4 for quarter spherical etc Tis 3 142 ris the distance between the source and the listening position Risthe room constant the room s ability to absorb sound i e the product of surface area and absorption coefficient 56 Page Copyright 2015 Rational Acoustics LLC All rights reserved We don t need to memorize these equations but understanding the relationship will help us trouble shoot problems For instance we see that Adding absorption will obviously reduce reverberation and increase the critical distance Alternatively we could reduce the effects of the reverberation by aiming highly directional loudspeakers into the audience but away from the walls and ceiling Conversely using less directional sources and or
46. el button then click the start gt button pop your balloon or fire your blank pistol or whatever give the system a few seconds to ring out and then click the stop button m to end the recording and display the results Saving Your Work When you measure an IR in Smaart does not automatically assume that you want to save it Sometimes it takes a few tries to get it right and we don t need a lot of old baggage piling up in the process When you have a measurement that you are happy with and want to save click in the File menu and select Save Impulse Response then select the directory where you want to store the file and give it a name Note that if you have cropped the file for display purposes using the Crop function only the displayed portion of the time record is written to file 49 Page Copyright 2015 Rational Acoustics LLC All rights reserved Recap Recommended Default Settings for Dual channel IR measurements The following are some recommended go to default settings for IR measurements in Smaart that should typically work well for most rooms Signal Type If you are able to use Smaart s Signal Generator x signal generator as your stimulus Signal Pink Noise On Levels J Show Peak Show RMS signal source then period T 1 Speech Weighted matched pseudorandom noise is Level 12 dB _ Band Limited a good all around choice for Period Random Si Pseudorandom Start Freq 100 0 signal type T
47. end of the time record The practical implication is that you may need to make the measurement time window a little larger than you would for a matched noise measurement to ensure these artifacts do not intrude on the reverberant decay slope As regards STI measurement IEC 60628 states that When using a sine sweep technique the distortion components that are inherent within the method shall be edited out or removed from the IR before calculation of the STI can be undertaken It is our opinion however that this requirement argues for the use of period matched noise rather than sweeps for STI measurement since the masking effect of high levels of distortion in lower spec announcement systems can significantly affect speech intelligibility and properly should be included in the measurement Distortion products arising from overdriving an excitation source 24 with a log swept sinusoidal test 30 i signal Pink Sweep 36 42 48 54 50 hi A 1 al I I 56 i a d Ly P 8 I TTT OTI ik hha ROT LETT TTT i yi ty TTT ty CTT 100 400 700 1000 1 30k 160k 1 90k 220k 2 50k 280k 3 10k 340k 3 70k 400k 4 30k 460k 490k 5 20k Figure 23 An impulse response measured using a log swept sine signal Pink Sweep showing harmonic distortion products from the excitation loudspeaker piled up at the end of the time record In this case the speaker being used to excite the room was overdriven and the distortion component was quite
48. equirement e Little or no averaging is necessary in fact averaging may actually increase noise artifacts e Measurement time constants can be kept to reasonable lengths e Small time variances become less of a concern e Subjectivity in selecting measurement parameters is reduced e The measurement system doesn t necessarily need to be connected to the system under test When using a known test signal the measurement system and the SUT can get their stimulus reference signals from two different sources and the measurement will still work You won t get an accurate propagation delay time without an audio feed from the signal source being used to excite the SUT but if you don t really need delay times this can be a very handy option Impulse Response Measurement Using Random vs Period matched Noise OE a Tarn Y 0 20 0 40 0 60 0 80 1 00 zs Fandom No Averaging Random 8 Averages Period Matched No Averaging Figure 22 Three indirect IR measurements of the same room taken from the same microphone position using effectively random noise vs period matched pseudorandom noise The period matched noise measurement in Green takes the same amount of time as the unaveraged random measurement in Blue but has much better dynamic range By repeating the random noise measurement eight times and averaging the results the measurement in Red we can greatly improve its signal to noise ratio however the measurement takes eight times long
49. er to perform Logarithmic Sweeps Logarithmic sweeps are called Pink Sweeps in Smaart When you select this signal type in the signal generator Smaart drops the IR data window without being told A data window in conjunction with a 39 Page Copyright 2015 Rational Acoustics LLC All rights reserved sweep signal would act as a filter on its frequency content since each frequency appears at only a single point in time during the measurement Sweeps can be used as a circular or aperiodic signal source If the Triggered by impulse response option is enabled in Smaart s signal generator the sweep signal is triggered by starting an IR measurement When you kick off the measurement Smaart will insert a short period of silence before the sweep in case there s any lag in starting the recording device then run the sweep and insert another period of silence afterward to let the SUT ring out If the Triggered by impulse response option is unchecked the sweep runs continuously when the generator is turned on In this case you would start the generator before starting the measurement as you would with other test signals A peculiarity of dual FFT based IR measurements made with logarithmic sweeps is that distortion products in the excitation loudspeaker SUT are washed out of the IR and show up as pre arrivals Because the DFT is a circular function these typically end up wrapped around past the beginning of the measurement and pile up near the
50. erial absorption coefficients 53 FT r 7510 OTS eege 53 Schroeder cut off FrEQUeiC yids EE 54 Practical measurement considerations ceeccccccssssscccccseeseccceeseeeeccecssaueecceesaueeeceessauseeesessaaseseeesaaaeeess 54 Appendix C Sound Reg ei Ee RE le 56 PIPE CTIVIEY FAC LON O E 56 Sound pressure level at the listening position 56 Biere ING OX DI EE 57 Qand Di TEUFeS ne Ra Ctl CO EE 57 CONVENTION EN Men ee E EE 58 PREMIO ON E 58 iii Page Copyright 2015 Rational Acoustics LLC All rights reserved EE EIERE ee eh eech 59 Appendix D Boundary eects EE 61 ACOUSTIC SMAI SOUC EE 61 Microphones near boundaries 61 Appendix E Typical Measurement ET Be CN 63 Stereo 22 AUdiO FO nsare N dau gndacctuiu nace saneanwageaccseonsmeeeicoeus 63 M ltkchannel Ga 63 Traditional USB FireWire Stand Alone Interfaces ccceeccccesseccccesececcesscceceesececeeececeeusesessuseseesunsces 63 NEtWWOrk FO srona O O TCP Ure reer ne gee en ere eer 64 iv Page Copyright 2015 Rational Acoustics LLC All rights reserved Introduction Scope and purpose of document Smaart 7 can operate in two distinct operating modes real time and IR short for impulse response mode Both real time and IR modes feature extensive capabilities for graphical analysis of audio data in the time and frequency domains however real time mode is primarily geared for analysis of streaming audio signals IR mode is purpose built for acoustical analys
51. es for the 125 and 250 Hz bands may be averaged together to get a Tow figure The average of the 500 Hz and 1 kHz band is called Twig When a single number figure is given for reverberation time it is assumed to be Tmig unless otherwise stated Smaart calculates these values for you automatically and displays them in the All Bands Table Dividing TLow by Tmia gives you the Bass Ratio Bass ratio quantifies the warmth of sound in a venue and is a particularly important parameter for concert halls The word Bass in this case refers to vocal or instrument bass registers and should not be confused with PA type sub bass frequencies Acceptable values are dependent on expectations A Bass Ratio of 1 1 1 25 would be regarded as good for fairly 23 Page Copyright 2015 Rational Acoustics LLC All rights reserved reverberant concert halls RT 60 greater than 1 8 seconds but the upper figure could be increased to 1 45 for less reverberant spaces As for what to look for in reverberation time results preferred reverberation times vary according to room size and purpose and the type of program material being presented In general you want to see shorter reverberation times for auditoriums classrooms theaters and cinemas ideally from about 0 4 0 5 seconds for smaller rooms up to 0 8 1 2 seconds for larger rooms Opera houses and mixed use performance spaces where both speech intelligibility and musical appreciation are equally important
52. es of a selected time range many go all the way to 0 so you generally want to position any peaks that you want to examine near the center of a selected range The Time O slider in the navigation pane can be used to move peak structures nearer to the center of the time window if they are too close to the edge to center up in the range Clicking in the right margin of the navigation clears a time zoom If you zoom way in on the IR and select a very narrow time range centered on the arrival of direct sound it s possible to see the magnitude response of loudspeaker without comb filters caused by early reflections at least at high frequencies In practice the usefulness of this strategy may be limited to how far away both the loudspeaker and microphone are from the nearest reflecting surfaces The frequency response of a DFT is limited by its time constant so you may find that by the time you squeeze the time window in enough to get rid of first order reflections you can t really see much detail in the frequency domain But it s something that people used to do quite a bit back in the days before lab measured anechoic response data for most professional loudspeakers became commonly available 26 Page Copyright 2015 Rational Acoustics LLC All rights reserved 80 Lin 2S Se eee rr 7 C ZP theater wav 80 E gt 1257 71 ms web 50 250 450 650 850 1 05k 1 25 145k 1 65k 185k 205k 225k 245k 265k
53. etary algorithm for IR saddle point estimation that has proven quite robust but it is not completely foolproof So it s still a good idea to check each band to make sure that you agree with the choices the software makes particularly if there are any large anomalies in the tail of the IR such as a prominent spike or distortion products piled up at the end of the record by a sweep signal Estimating Reverberation Time by Reverse Integration of the Squared IR e Arrival of Direct Sound M 5 dB Point Lr1 Impulse Reponse Reverberant Decay Range Reverse Integration from Saddle Point 35 dB Point Lr2 T 20 Decay Range AN D wi E m amp T a EEEE Arbitrary Reverse Sea e o Ine grat ion Saddle Point Reverberant Decay Slope EEEE E H oise Floor 0 4 Time Sec Figure 8 Estimating Reverberation time by reverse integration of the impulse response Reverse integration of the IR from the saddle point the approximate point where the reverberant decay slope meets the noise floor of the measurement provides a very good estimation of reverberant decay time Starting the reverse integration from an arbitrary point such as the end of the file may result in overestimation of decay time Evaluation Ranges EDT T20 T30 Because it is rarely possible to actually measure a full 60 dB of reverberant decay in acoustical systems reverberation is typically evaluated over a smaller range The starting
54. ey are digitized and distributed via the digital snake Using Dante Network Protocol as an example in a setup such as this the computer operating Smaart sees Dante Virtual Soundcard as an audio I O device All routing for the soundcard is done within Dante s Controller application The computer running Dante Controller in this example is also running or hosting the Smaart application has the ability to see any signal connected in the network With the ability to analyze any signal on the network this can provide for some extremely flexible measurement configuration possibilities 65 Page Copyright 2015 Rational Acoustics LLC All rights reserved
55. f the source the volume and reverberation time of the room and the level of background noise A longer form of the calculation suitable for direct measurement was later developed and found its way into various acoustical measurement platforms including Smaart The long form uses direct vs reverberant energy usually based on C20 or C10 in place of Q distance and volume ALCons can be calculated for any frequency range but it is conventionally regarded as most meaningful in the octave band centered on 2 kHz as this is where most of the energy in consonant sounds is found Advantages of ALCons include the fact that in its original form it made estimation of intelligibility possible for sound system designs not yet installed in rooms not yet been built without the aid of acoustic modeling programs not yet available in the 1970s and 80s In its long form it could be directly measured in existing installations based on ETC or IR data produced by the TDS and MLS FHT measurement systems that were prevalent before the computing power needed to calculate FFTs large enough for room acoustics work became widely accessible Its main disadvantages are its reliance on assumptions that the sound field is statistically well behaved and without audible echoes which may or may not be true Qualitative Thresholds ALCons is upside down relative to other intelligibility metrics Smaller numbers mean better scores Anything less than 5 is considered excellen
56. filter can be set to any value between O Hz and one half of the Nyquist frequency equal to one half of the sampling frequency for the currently selected audio sample rate Frequency Display Settings e Smoothing echoes the setting of the Smoothing control in the upper right corner of the Frequency graph in IR mode e The Magnitude range controls are useful for setting a specific decibel range for the Frequency graph in IR mode however you can also resize the range using the keys or by right click and drag mouse zooming as you can with other graph types in Smaart Spectrograph Settings e The FFT Size and Overlap controls echo the settings of the controls found in the upper right corner of the Spectrograph display in IR mode Together they determine the time resolution of the spectrograph e Grayscale plots the spectrograph using varying shades of gray instead of color to represent magnitude e Data Window sets the data window function used in calculating the individual FFTs used to create the spectrograph display You can leave this set to Hann unless you have some good reason to change it e Dynamic Range echoes the settings of the slider control widgets found on the left edge of time domain and spectrograph displays on IR mode The Spectrograph scales its color or grayscale spectrum to the range between the Min and Max values and plots decibel values above the Max thresholds in white and below the Min in black Histogram Settings
57. ge gg a Se _amamemenamemamamamamamamnanamed 40 40 40 80 80 80 290 360 430 500 570 640 710 780 290 360 430 500 570 640 710 780 290 360 430 500 570 640 710 780 Spectrograph V Spectrograph V Calc l Calc FFT 2k FFT 2k FFT 2k Overlap 50 Overlap 75 Overlap 99 260 310 360 410 460 510 560 610 660 710 760 260 310 360 410 460 510 560 610 660 710 760 260 310 360 410 460 510 560 610 660 710 760 Figure 16 The 2K FFT example from Figure 14 shown with 50 75 and 99 overlap left to right 30 Page Copyright 2015 Rational Acoustics LLC All rights reserved Spectrograph Dynamic Range White The dynamic range of the Spectrograph is controlled by two arrowhead pi a e Max dB shaped widgets that appear on the left edge of the Spectrograph chart RN These controls are echoed on the Log IR and ETC displays where they relate MASSON directly to the decibel levels on the graph The upper of the two widgets 42 Dynamic sets the maximum Max threshold the lower one sets the minimum Min tin The Spectrograph scales its color spectrum between these two extremes 60 Any FFT bin whose magnitude exceeds the specified maximum is mapped S ei Min dB to the color white Values falling below the minimum are mapped to black anne Spectrograph Analysis of an Acoustical Impulse Response Long time Smaart users may actually recognize the impulse response shown in Figure 18 It was made using the file room wav that was distribu
58. gth O Frequency amp Note ID Hei Cursor Time Readout General Options Pertaining to IR gl Miiliseconds O Miiliseconds amp Distance Measurement Cursor Behavior Free Cursor Tracks Data wi Locked Cursor Tracks Data Cursor Time Readout a ee A e Speed of Sound The cursor time readout setting applies to the O Eo E both IR Mode and the Live IR graph on Transfer e S Clock Function displays e e Milliseconds displays time coordinates OK Cancel Apply and relative time differences in 11 Page Copyright 2015 Rational Acoustics LLC All rights reserved milliseconds only e Milliseconds amp Distance displays time coordinates as milliseconds and equivalent distance based on the Speed of Sound settings see below Speed of Sound The settings in this section determine the speed of sound that Smaart uses for calculating equivalent distances for time coordinates and also whether distances are displayed in feet or meters It can also serves as a handy speed of sound calculator any time you need to know the speed of sound for a given air temperature e Use Meters Celsius when this option is selected Smaart displays distances in meters and the temperature used for calculating soeed of sound in degrees Celsius Otherwise Smaart displays distances in feet and uses degrees Fahrenheit for temperature e Speed of sound unit sec and temperature At elevations where humans wou
59. iagrams Excitation Level The rule of thumb for setting the excitation level for IR measurements is that you would like to be able to get at least 40 50 dB above the background noise level In reverberation time measurements we evaluate reverberant decay over a range starting 5 dB down from the arrival of direct sound normally the highest peak in the IR and extending down another 20 or 30 dB from the start point A 30 dB range is preferred but 20 dB is OK if you can t get 30 Either way the lower end of the range needs to be at least 10 dB above the noise floor of the IR measurement When you add that all up you re looking for a minimum of 45 dB of dynamic range for a 30 dB evaluation range and at least 35 dB for a 20 dB range and that s in a perfect world with no noise artifacts from the measurement process itself In the real world adding another 5 to 10 dB on top of that would be a definite nice to have unless of course that would drive the system into distortion or blow something up To figure out how loud you need to be you can simply measure the background noise level We are looking for a relative relationship so you don t even really need to be calibrated for SPL unless perhaps you plan on doing an STI measurement Just set the sound level meter in Smaart to Slow SPL and watch the meter for ten or twenty seconds with no output signal running to get a feel for the baseline noise level then start the signal generator at a l
60. icability of early to late energy ratios as good predictors of subjective speech intelligibility and RT60 as a somewhat rougher gauge There also exist some purpose built metrics created specifically for predicting speech intelligibility Two of these ALCons and STI can be calculated from the impulse response of an acoustical system STI and STIPA The Speech Transmission Index STI has emerged over the last decade or so as the go to metric for objective estimation of speech intelligibility in acoustics STI is a relative of the Articulation Index Al which is based on speech to noise ratios across a wide range of frequencies But unlike Al and SII STI works well for reverberant environments in addition to communication systems Rather than estimating intelligibility based on direct to reverberant or signal to noise ratios STI starts with the concept of speech as a carrier frequency sound from our vocal cords that is modulated by very low frequency fluctuations as the speaker s mouth and tongue move and change shape to form words Looking at Figure 19 a segment of actual human speech it s not hard to see how someone might arrive at that conclusion Figure 19 A recording of a male voice reciting Joe took father s shoe bench out The basic idea is that most of the information in speech is carried in these low frequency modulations and anything that reduces the depth of the modulations must negatively affect speech intelligibi
61. ifference between the Log IR and the ETC is pretty striking see Figure 7 To scale your display to look like Figure 7 press the plus key on your keyboard a few times to zoom in on the magnitude range then use the up down arrow keys to move the range up and down Note that when using peak locations to find delays the ETC can sometimes give you a slightly different answer than the Log IR because of the way it effectively interpolates between peaks in the IR If you look at the smaller peak in the ETC at about 124 ms in Figure 7 you can see that it falls in between two lobes in the Log IR We have found that the ETC can be more effective than the Log IR tool for finding subwoofer delay times But that is better done in real time mode using the ETC on the Live IR in conjunction with the frequency domain transfer function displays where you can see phase as well as magnitude and watch changes happening in real time as you adjust processor settings That s a little beyond the scope of this document 18 Page Copyright 2015 Rational Acoustics LLC All rights reserved HAN 200 220 20 40 60 80 100 120 140 160 180 200 220 240 Figure 7 A comparison of the Log IR and ETC graphs in Smaart for the 125 Hz octave band in room wav Bandpass Filtering Up to now we have mainly been looking at the broadband IR but quite a lot of acoustical analysis is conventionally done using octave or sometimes 1 3 octave bands especially
62. igital full scale The dBA SPL Slow Max 42 8 labels above and below the large numeric value in the center of the display function as menus that enable you to select the input channel driving the display and the type of measurement you wish to perform When operating as an SPL or peak meter the Max value can be reset by clicking the red circle e in the lower right corner of the display In LEQ mode it changes from red to green e once the buffer for the rolling LEQ average is fully populated and clicking it will clear the buffer and restart the LEQ Using the K hotkey displays a clock with the system time in the SPL meter window Clicking anywhere in center portion of the meter opens the Sound Level Options dialog window where you will find additional options for configuring and calibrating the meter 5 Data Display Controls The first group of controls in the control strip on the right side of the All Bands T 0 RealTime IR mode window is devoted to data display functions Starting from Se ES Schroeder the top left of the screen clip on the right LZ e The All Bands button opens the All Bands table where all of the quantitative acoustical metrics that Smaart can calculate are found e Clicking the T60 displays the level marker widgets used for calculating reverberation time and early decay time on Log IR or ETC plot s e The Real Time button exits IR mode and takes you back to real time frequency domain measurement m
63. in graphs Lin Log or ETC displayed in the main graph area 2 Main Graph Area The larger lower portion of the graph area can be divided into one or two panes by clicking the buttons labeled with rectangles in area 5 Each pane can host any one of the six available main graph types e Lin Log ETC time domain views e Spectrograph frequency and level vs time e Frequency spectrum e Histogram bar chart of quantitative acoustics values by octave or 1 3 octave band We will discuss each of these display types in the next section The graph type for each pane is selected by means of the drop list control in the upper left corner of the pane Some main graph types also have E Be Gin additional selector controls in their upper right corner that control display 18 ETC options specific to that graph The two arrowhead shaped widgets to the 24 Frequency 30 Spectrograph left of the list of graph types appear on all time domain and spectrograph plots They control the dynamic range for the spectrograph e SE As you probably noticed the main graph area and the navigation pane are blank when you first enter IR mode You won t see any data in the graph areas until you record a measurement or load an impulse response from file File menu gt Load Impulse Response Smaart can open and analyze wav and aiff files containing any type of audio data but IR mode is purpose built for analyzing impulse responses There s no multi
64. in the first n milliseconds of the arrival of direct sound inclusive to the energy in the remainder of the reverberant decay period The two most commonly used are C50 and C80 which use at the 50 or 80 milliseconds respectively as their split times The result of the comparison is expressed as a decibel ratio POOR FAIR GOOD SPEECH Ca t rtt e t rF r ioter ee e e 12 10 8 6 4 2 0 2 4 6 8 10 12 14 16 18 dp MUSIC Ca 4H ORGAIN SYMPHONY OPERA ELECTRONIC INSTRUMENTS TEST FREQUENCIES SPEECH 0 5k 1k 2k amp 4kHz octave band Ce values intelligibility weighted amp summed MUSIC 0 5 1k amp 2kHz octave band Cg values averaged Figure 10 A scale for interpreting C50 and C80 measurement results for speech and music 24 Page Copyright 2015 Rational Acoustics LLC All rights reserved Shorter split times such as 35 or 50 ms are regarded as better predictors of speech intelligibility C80 is more useful for music In terms of what kinds of numbers to look for Gerald Marshall provided the table shown in Figure 10 in the in a 1996 Journal of the AES article titled An Analysis Procedure for Room Acoustics and Sound Amplification Systems Based on the Early to Late Sound Energy Ratio For the speech intelligibility side of the graph Marshall used a weighted average of the 500 Hz to 4 kHz octave bands with the following weights assigned to each band 15 for 500 Hz 25 for 1 kHz 35 for 2 kHz and 25 for 4 kHz Others have
65. ingle rack space format makes portability in a rack case or in your backpack equally viable In addition to 63 Page Copyright 2015 Rational Acoustics LLC All rights reserved fielding more microphones having more inputs enables you to tap into electronic measurement points as well ee ei ss o Mic 3 System Control Speakers Console Out DSP Out Smaart Signal Gen Out USB or FireWire cr O00000 bettel ss ss a as eo Multi Channel Audio 1 0 Above we have three measurement microphones as well as electronic measurement points pre system processing pre speaker system and a hard wired loopback pre console In this example the Console Out and DSP Out can be used as reference signals that let you isolate the response of the system post console or post DSP or as measurement signals e g for graphing an EQ response in Smaart Network I O Preamp Returns elle KEE KKK KKK KKK 68698600 System Amplification Audio Components Network Switch System Processing The ability to fully digitize your audio stream from the stage pre amp all the way to the speaker system has become a reality as of late In some professional systems nowadays the only analog signals present 64 Page Copyright 2015 Rational Acoustics LLC All rights reserved are those from the microphones to the stage conversion box where th
66. is In IR mode Smaart can work with time records several seconds in length loaded from a file or actively measured and can calculate metrics used in room acoustic and estimate speech intelligibility from IR data This document is intended as a practical guide to the basics of measuring and analyzing an acoustical impulse response IR using Rational Acoustics Smaart We intend to focus on the practical as much as we can without spending a lot of time on theory or mathematics Our main objective is to help Smaart users become better acquainted with the process of selecting measurement parameters and microphone positions to get a good IR measurement and hopefully extract some useful information from the measurement results 1 Page Copyright 2015 Rational Acoustics LLC All rights reserved 1 What is an impulse response In the most basic terms an impulse response IR can be defined as the time domain time vs amplitude response of a system under test SUT to an impulsive stimulus The word system in this case could mean something as small as a microphone or a single transducer something as simple as a single filter on an equalizer Or it might mean something as big as a concert hall or sports arena as complicated an entire sound system or a combination of the two Smaart users of course are most often concerned with sound systems and their acoustical environments In the context of acoustical analysis you might also thi
67. ival that is 90 or 270 shifted shows as a zero crossings thereby making a single arrival that is soread out over time and phase appear to be multiple arrivals The Energy Time Curve also called envelope of the impulse response represents the magnitude of the energy arrival over time by effectively ignoring phase The textbook description is the real impulse response combined with its Hilbert transform a copy of itself that has been rotated 90 in phase In practical terms the summation of the two tends to fill in zero crossings seen in the Log IR producing a signal that can be a lot easier to look at than the Log IR by virtue of being less squiggly At higher frequencies the Log IR and ETC may look very similar both are plotted on a logarithmic magnitude scale but the ETC is particularly useful sizing up the arrival of direct sound at low frequencies 17 Page Copyright 2015 Rational Acoustics LLC All rights reserved IR Peak r ETC Peak 100 0 Actual Magnitude dB FS 0 r Linear Amplitude Full Scale r maximum dynamic range given bits sample 100 Time gt 0 ms E Linear Impulse Response Logarithmic View of Impulse Response E Energy Time Curve ETC with Logarithmic Amplitude Scaling Figure 6 A comparison of the ETC and the impulse response with linear and logarithmic amplitude scaling If you zoom in on the first 250 ms of room wav and switch to the 125 Hz octave band the d
68. ively choose for frequency domain transfer function measurements of a sound system Selecting Measurement Parameters Once you have done all the groundwork of determining your source and measurement positions and figuring out which measurement technique to use the part of the measurement procedure that directly involves Smaart is actually pretty easy Basically you just need to select your measurement parameters turn on the signal generator or other stimulus signal source and kick off the measurement The two main things you need to concern yourself with at that point are the stimulus level and the measurement duration which will be some combination of the FFT size and the number of averages 46 Page Copyright 2015 Rational Acoustics LLC All rights reserved Input source If you already have one or more Transfer Function measurements configured and will be using one of those to make your measurement use the Group and TF Pair selectors shown in to select the one you want To create a new TF pair click the little hammer and wrench button next to the Group selector to open the Measurement Config window then click the New TF Measurement button see Figure 26 This pops up another dialog where you can select the input device and channels that you want to use and give the measurement pair a name If you are unfamiliar with how to set up your measurement system for transfer function and dual channel measurements Appendix E has some example setup d
69. l of the other markers Ideally this will roughly correspond to the saddle point in the impulse response The magnitude coordinate is used to estimate the 22 Page Copyright 2015 Rational Acoustics LLC All rights reserved level of the noise floor of the measurement and the Lr2 marker needs to be at least 10 dB above that As of version 7 5 Smaart does a pretty good job of placing the Ln marker most of the time However it may still benefit from a human touch in some cases particularly if the dynamic range of the measurement is marginal or there are significant distortion artifacts from a swept sine measurement or any other prominent anomalies in the noise tail of the IR being analyzed When the level marker widgets for reverberation time are visible a block of vital statistics also appears in the upper right corner of the plot These include the 60 dB reverberation and early decay times RT60 and EDT and the time and level differences between three pairs of markers The Ld Le level difference should always be 10 dB Lr1 Lr2 should be either 20 or 30 dB this number is convenient for checking your work if you end up adjusting Lr2 is by hand The Ld Ln delta is interesting also as it gives you the dynamic range of the measurement D R stands for direct reverberant ratio It is an early to late energy ratio that gets its split time from the time coordinate of the Le marker Saving your work If you do end up adjusting any of the Level marke
70. lay pane s to the full IR time record 60 Liny The selector control in the upper right corner of the navigation pane selects 2 the graph type to be displayed in this area The navigation pane is limited to 40 z 80 5 25 50 Zi 7 Page Copyright 2015 Rational Acoustics LLC All rights reserved time domain graph types only Lin Log or ETC The Crop button in the lower right corner of the navigation pane can crop a file for display D purposes to show only the selected time range a very useful feature when working with IR measurements with long noise tails Cropping is non destructive and can be undone clicking the Crop button again on a cropped measurement restores the full extent of the original time record however if you save the IR to a file while cropped the cropped version is written to file 60 Lin 40 The little arrow shaped widget positioned in the lower left corner of the 0 navigation pane circled in red in the screen clip show the right is the Time O Si marker When you record a dual channel impulse response measurement in ES Smaart this marker is automatically set to match the reference signal delay time if you are familiar with real time transfer function measurement it s analogous t the center point of the Live IR For single channel measurements or file based data it is set to the beginning of the time record Dragging it to the left or right moves the time zero point for all time doma
71. ld be comfortable breathing the speed of sound is mainly a function of temperature and so the two input are linked Changing the temperature setting automatically recalculates the corresponding speed of sound and vice versa Impulse Response Options Most of the controls in the Impulse Response options dialog tab echo the settings of the controls in the main window We have already discussed those in some detail so we will concentrate here on the few that don t Time Domain Display Settings e FFT Size and Averages echo the settings of the on screen controls for dual channel IR measurements that we talked about previously in topic 7 Options x i General Spectrum Transfer Function Impulse Response Pelay Zoom Time Domain Display Settings FFT Size 128k Show IR Peak V Averages None v AlCons Split Time ms 10 Overlap 0 Mag Threshold dBFS o Filter Settings High Pass Filter V Frequency 8 00 Frequency Display Settings Magnitude Range dB Smoothing 1 6 Oct X Max 36 in 54 Spectrograph Settings FFT Size 2k Data Window Hann Overlap 99 Dynamic Range dB FS Max 0 Min 23 Grayscale Histogram Settings Plot As Une OK Cancel Apply e Overlap for Averaging not to be confused with overlap for the Spectrograph display When Overlap is set to any value other than zero each successive measurement going into an averaged d
72. les at the currently selected sampling rate Averages Avg sets the number of successive IR measurements to average together to improve the signal to noise ratio of dual channel measurements using random test signals For deterministic IR measurements made using period matched signals the number of Averages is normally set None 8 Input Source Selector Dual channel IR measurements in Smaart are essentially transfer _ Group Defaut TF 5 TF Pair USB Device function measurements with the addition of an inverse Fourier transform at the end Any of the reference and measurement signal pairs that you have set up for real time transfer function measurements may be used for IR measurements For single channel IR measurements see 9 below only the measurement channel is recorded If you have more than one Transfer Function measurement group the Group control is used to select the group containing the measurement that you want to use 9 Live Measurement Controls The live measurement controls in IR mode are analogous to a transfer al H ye DH gt a i Cont function measurement engine in real time mode with a couple of i f Read extra twists Starting from the top left the buttons marked with a siaki triangle gt and a square m start and stop a measurement The button Mi m 150 Rs vc labeled with a circle e works like the record button on a tape deck or digital recorder but in this case it s also a
73. lity The real advantage of this approach is that STI ends up being sensitive to just about any factor that works to degrade speech intelligibility in a sound system and or a room including noise excessive reverberation distortion and audible echoes The basis for STI is the modulation transfer function MTF which quantifies the depth of modulation in the received signal relative to the transmitted signal at specified frequencies The modulation transfer 32 Page Copyright 2015 Rational Acoustics LLC All rights reserved function can be measured directly using specialized speech like test signals or calculated indirectly from the impulse response or ETC of a system under test In either case it is measured over a range of seven octaves from 125 Hz to 8 kHz at fourteen modulation frequencies per band The modulation frequencies range from 0 63 Hz to 12 5 Hz in 1 3 octave intervals Modeling Modulation Loss Due to Reverberation and Noise Envelope of Amplitude Modulated Transmission Signal wee Modulation Frequency Interfering Reverberation and Noise wee Received Modulation Signal with Modulation Loss Figure 20 Estimating speech intelligibility based on modulation loss in an amplitude modulated signal The black line is the modulation signal for the transmitted signal The red line is the modulation signal of the received transmission The difference between the two tells you the loss of modulation through the transmission chan
74. ll get to that in the next section is considered to be early decay Reverberant decay is conventionally measured over a range from 5 dB below the level of direct sound down to a point 30 dB below that on the integrated IR or 20 dB in a pinch Noise Floor In theory the reverberant decay phase of the IR continues forever as an ideally exponential curve that never quite reaches zero In practice it reaches a point relatively quickly where we can no longer distinguish it from the noise floor of the measurement Noise in an IR measurement can come from several sources including ambient acoustical noise and electrical noise in the SUT and the measurement system quantization noise from digitizing the signal s for analysis and process noise from DSP processes used for analysis 4 Page Copyright 2015 Rational Acoustics LLC All rights reserved Uses for impulse response measurement data Delay Time Measurement The most common use for impulse response measurements in Smaart is in finding delay times for signal alignment in transfer function measurements and for aligning loudspeaker systems Each time you click the delay locator in Smaart an IR measurement runs in the background In this case all we really care about is the initial arrival of direct sound which is typically so prominent that you can pick it out with high confidence even when signal to noise ratio of the IR is poor so we don t even bother displaying the results Smaart simply sca
75. matched noise as your excitation signal you can set the delay time to 0 if you don t already have it set for your selected signal pair If you do then there s no harm in leaving it alone If you are using a random noise source and don t already know the delay time for your signal pair run the IR measurement once to find it then click the button to set it then run the measurement again 50 Page Copyright 2015 Rational Acoustics LLC All rights reserved 51 Page Copyright 2015 Rational Acoustics LLC All rights reserved Appendix A Applicable Standards and Further Reading Several of the techniques procedures and quantities discussed in this document are the subject of ISO and IEC standards We highly recommend reading those standards rather than relying on our interpretations and synopses The main one is ISO 3382 Acoustics Measurement of room acoustic parameters parts 1 and 2 e Part 1 Performance Spaces ISO 3382 1 seems to be geared more toward concert halls opera houses and theaters and contains a quite a bit of information on acoustical quantities other than reverberation times not found in e Part 2 Reverberation time in ordinary rooms ISO 3382 2 We note that there is quite a bit of overlap between the two In fact the rationale for making it a two part standard doesn t seem immediately obvious to us but if you didn t want to buy both parts part one is the more comprehensive of the two Both
76. n your keyboard or click the mpulse button that appears in the upper right corner of the main window in real time mode just below the signal level sound level meter You will find yourself confronted with a screen like the one below The colors may be darker but the layout is the same On the right side of the window is a vertical strip of controls The rest of the window is devoted to the graph areas and cursor readout Smaart v7 File Options Mode View Zoom Cursor Help ey Lett 3 Cursor Readout e Ko FES 4 Sound Level Meter 58 3 dB SPL Slow Max 58 5 175 200 225 250 275 O Data Display Controls Impulse Response O Signal Generator FFT 16k 341 ms Avg None FFT size and Averaging Group DefaultTF L t Main Graph Area TF Pair USB Device Input Source Selector Live Measurement Controls aN lt oe Es Filters Octave a do Bandpass Filters 1 Navigation pane The small time domain display in the upper part of the graph area is used for navigation and is always visible Right clicking and dragging across this Ctrl click and drag on Mac panel selects a specific time range for display on the larger time domain charts The full IR time record remains visible in the navigation pane when you are zoomed in unless you use the crop function Clicking anywhere in the left margin of the plot clears the zoom range and returns any time domain graphs in the main disp
77. nel due to interference from reverberation in the case of an acoustical system and noise The current IEC standard on STI 60268 16 Edition 4 0 2011 06 says that when measuring STI indirectly from an impulse response The duration of the impulse response shall not be less than half the reverberation time and at least 1 6 s to ensure a reliable calculation of the modulation indices for the lowest modulation frequency of 0 63 Hz However to us this seems to overlook the fact that frequency bins in a DFT are linearly spaced meaning two lowest bins are always an octave apart whereas the modulation frequencies for STI are on 1 3 octave intervals If the DFT time window is 1 6 seconds then the two lowest bins are at 0 63 and 1 25 Hz whereas the first three STI modulation frequencies are at 0 63 0 8 and 1 0 Hz This is to say that hitting all of the lowest STI modulation frequencies would require a measurement window significantly longer than 1 6 seconds We experimented and found that 5 seconds was a sweet spot producing data points very close to all of the STI modulation frequencies That is the reason for the 5000 ms DFT size introduced in Smaart 7 5 for IR measurements The DFT size in samples depends on the sampling rate you are using For example at 48k sample rate 5000 ms works out to 240 000 samples 33 Page Copyright 2015 Rational Acoustics LLC All rights reserved IEC 60268 16 qualifies IR based STI measurement techniques for
78. ning to IR Measurement eseessesreeseeressresreeseeresrresreeseeresrresreeseeresreessersseresreess 11 Jet ei Response Opos E 12 3 Analyzing le le EE Response Datla WEE 14 i Page Copyright 2015 Rational Acoustics LLC All rights reserved Hie Ee a ENEE 14 Logarithmic Lime DOMalM BET 14 Linear Time Raul BEIEN e 15 Energy Time Curve ETC sesssccsessssccevsavavonsacviesvsanesdsavecedusardsvevsanecesviwetov sd ais svonsecdsVasevgssaaeiwessvedeaseraneevers 17 SET tele Ge e 19 Discrete e de EE 19 Reverperation Tl EE 20 Reverse Time INTC FAO EE 20 Evaluation Ranges LEDs 120 T30 EE 21 Reporting Results for Reverberation Time ccccccsssccccesscccceesececeesecceeescceseeececeuneceseueeceeseuecetseneees 23 Eariy tokate ENeCrey Ee E 24 Clarity Ratios C35 C50 E EE EE 24 MME AISCORFAM Display iora eege 25 NEE lee E Ke EE 26 FREQUENCY Reeg He EE EE 26 TINE SPECEOS FAD BEE 27 Spectrograph Time and Frequency Resohutton 29 SHECTFOST ap DYNAMIC Ee TE 31 Spectrograph Analysis of an Acoustical Impulse Response ssesssssssssssrrnserrssrrrssrrssrresrrrsrrreerrrerrreerreses 31 SHEECEM IMTOMPONOINTY EE eege ee erg 32 BEER EE 32 POU ONS EE 35 Measuring an Acoustical Impulse Response ansiar nsa AEA EARE EES 36 What are We Measuring ANG WY EE 36 Direct vs INGIFectIR MES DEET green ease 36 Direct IR Measurement Using an Impulsive Stimulus 36 Indirect Dual Channel IR Measurement ccccccseccceeecccenscecensecescsseueceeeu
79. nk of an impulse response as the acoustical signature of a system The IR contains a wealth of information about an acoustical system including arrival times and frequency content of direct sound and discrete reflections reverberant decay characteristics signal to noise ratio and clues to its ability to reproduce intelligible human speech even its overall frequency response The impulse response of a system and its frequency domain transfer function turn out to be each other s forward and inverse Fourier transforms va Bang 4 am Reflected Sound Direct Ei Sound Figure 1 An acoustical impulse response consists of sound from an excitation source arriving at a measurement position by multiple pathways both direct and reflected Here we see the path of direct sound from the source to the microphone in red followed by a first order reflection in blue a second order reflection in green and higher order reflections in gray Later arrivals tend to pile on top of each other forming a decay slope An acoustical impulse response is created by sound radiating outward from an excitation source and bouncing around the room Sound traveling by the most direct path a straight line from the source to a measurement position arrives first and is expected to be the loudest Reflected sound arrives later by a multitude of paths losing energy to air and surface absorption along the way so that later arrivals tend 2 Page Copyright 2015 Rati
80. ns for the highest peak and assumes that to be the first arrival and most of the time that works very well Occasions where automatic delay measurements might not work well include measurements of low frequency devices or any case where you re trying to measure a directional full range system well off axis in a location where a prominent reflection can dominate the high frequencies In the latter case it s possible for reflected HF energy to form a higher peak later than the arrival of direct sound requiring you to visually inspect the IR data to find the first arrival Reflection Analysis Another common use for IR measurements is in evaluating the impact of problematic discrete reflections Reflected sounds can be beneficial or detrimental to a listener s perception of sound quality and or speech intelligibility depending on a number of factors These factors include the type of program material being presented generally soeech or music the arrival time and overall level of the reflected sound relative to the level of direct sound and the frequency content and the direction from which they arrive As a general rule the later they arrive and the louder they are relative to direct sound the more problematic they tend to be Reverberation Time T60 RT60 Reverberation time is kind of the grandfather of quantitative acoustical parameters First proposed by Walter Sabine more than 100 years ago T60 or RT60 reverberation time is th
81. o turn on this option open the signal generator Cycle 512k Wl Drop IR Data Window Stop Freq 1000 0 control panel select Pink Noise as the signal type then tick the Device Wave Speakers D Sm Main Left e Aux Right boxes labeled Pseudorandom and OK Gancel Drop IR Data Window FFT 128k 2730 MS FFT size and Averaging Avg 2 128K makes a good default setting for FFT size At 48k sampling rate that gives you almost 3 seconds of time window You would generally have to be measuring in a pretty huge space to need more than that but it s not ridiculously long for smaller venues For averages a setting of 2 is a good when using period matched noise If you have to use a random signal source for some reason up the number of averages to 8 or maybe 16 if you are working in a noisier environment We are assuming 0 overlap for averaging Excitation Level If you need to measure reverberation time then your excitation level needs to be a minimum of 45 dB above the background noise level for T30 preferred or at least 35 dB above to get T20 For most other purposes any excitation level that is comfortably above the background level should be fine Input Levels When using random or pseudorandom noise signals 10 to 12 dB or so is the preferred input level for any kind of measurement in Smaart including IR measurements 12 dB is the point where the input levels in Smaart turn yellow Delay Time When using period
82. od at alte processing them They may be useful or detrimental depending on such factors 8 kHz as their arrival time and loudness relative to direct sound the two biggies their frequency content and even the angle they arrive from Discrete reflections can cause audible problems ranging from coloration timbre change to image shift to audible echoes But trying to figure out which reflections are friend or enemy by looking at squiggly lines on a computer screen can be a bit of a dicey prospect 19 Page Copyright 2015 Rational Acoustics LLC All rights reserved Short reflections arriving within the first 30 milliseconds or so after the direct sound at relatively high levels are notorious for producing comb filters that muck up our real time frequency domain analysis but humans actually find them beneficial enhancing the intelligibility of soeech and the clarity of music Outside that early integration window reflections can still contribute to subjective impressions of presence warmth spaciousness etc But the rules are a little different for soeech vs music Individual broadband reflections arriving at 95 ms or more can destroy speech intelligibility and also make life difficult for presenters and performers if they reach the stage At that point you re getting into the high end of the syllabic rate for very fast human speech and people find it disorienting to be hearing the last syllable they spoke or sang as they are attempting
83. ode In real time mode this changes to the Impulse button that will bring you back to IR mode e The two buttons labeled with rectangles divide the main plot area 2 into one or two graph panes 1 rectangle 1 pane 2 rectangles 2 panes 9 Page Copyright 2015 Rational Acoustics LLC All rights reserved e The Schroeder button displays a reverse time integration curve on Log IR or ETC plots 6 Signal Generator The Signal generator controls work the same way in IR mode as in Pink Noise 48 dp real time mode The leftmost button displays the currently selected i signal type and turns the generator on and off The buttons on the right increase decrease the output level The center area displays the current output level and doubles as a button to open the signal generator control panel also accessible by selecting Signal Generator from the Options menu if you click it with your mouse We will be talking more about signal generator options later on as there are a few that are of particular interest in the context of IR measurements 7 FFT size and Averaging Controls The FFT size and averaging controls together determine the measurement FFT 64k 1365ms duration for dual channel IR measurements Notice that for each FFT size the D Avg None time constant is given along with the FFT size in samples The FFT time constant also called the time window is the amount time it takes to record the required number of samp
84. oes the other time domain Lin Log and ETC and Frequency graphs in IR mode but moving Time 0 cropping the time record or filtering the IR will clear the spectrograph and require clicking the Calc button again You can resize the spectrograph and move its range using the arrow keys or right click and drag on the plot to zoom in on a selected range as you can with any other graph in Smaart Also as with other graph types in Smaart clicking in the left margin of the plot after zooming in will clear the zoom and return the plot to its previous x y range Spectrograph V Calc FFT 1k Y Overlap 0 Spectrograph Calc FFT 512 Overlap 0 J Time Resolution 21 3 ms gt Time Resolution 10 7 ms gt ef mMM E mani an Tit a as a at a Pit a ett SET d Sp WH EE ess CH Frequency Resolution 94 Hz dere Resolution 47 Hz 20 70 120 170 220 270 320 370 420 470 520 570 620 670 720 770 820 870 920 970 20 70 120 170 220 270 320 370 420 470 520 570 620 670 720 770 820 870 920 970 Spectrograph Calc Spectrograph Calc FFT 2k V FFT 4k V z ka Overlap 0 Overlap 0 Time H Time Resolution 85 3 ms gt Frequency Resolution 23 Hz Resolution 12 Hz 20 70 120 170 220 270 320 370 420 470 520 570 620 670 720 770 820 870 920 970 20 70 120 170 220 270 320 370 420 470 520 570 620 670 720 770 820 870 920 970 Figure 14 Spectrograph time and frequency resol
85. oints Direct IR Measurement Using an Impulsive Stimulus The most intuitive way to measure the impulse response of a system would be to use an impulsive stimulus of some kind and simply record what happens And in fact people have been doing just that for decades The advantages are that you do not need a sound system or even a measurement system to 36 Page Copyright 2015 Rational Acoustics LLC All rights reserved perform the measurement All you really need is some way to make a loud bang and some way to record it The main problem with this approach other than the fact that it doesn t tell you anything about an installed sound system if applicable is the scarcity of really good impulsive stimulus sources An ideal impulsive stimulus would be a perfectly instantaneous perfectly omnidirectional burst of energy containing equal proportions of energy at all audible frequencies In the time domain it would appear as a single vertical spike no more than one sample in width In the frequency domain it would produce a perfectly flat magnitude and phase trace When evaluating the response of your system under test SUT to this ideal stimulus you could then confidently assume that anything you saw in the IR that wasn t an instantaneous spike or anything in the frequency domain that wasn t a flat line must be the response of your system If you loaded the wave file JsamplePulse wav to look at the frequency response of Smaart s
86. om an excitation source Admin 2 Vp Eq 1 where V is the volume of the room in cubic meters c is the speed of sound in meters sec is estimated reverberation time in seconds But we did promise to keep the math to a minimum so Figure 25 provides a handy graph of this relationship to go with it 43 Page Copyright 2015 Rational Acoustics LLC All rights reserved Room Volume in Cubic Feet 100k 300k 500k 700k 900k 200 mmm RT60 1 sec z mm RT60 2 sec We z E e A s m RT60 3 sec E 150 3 m 3 CG 3 1235 4 Ki S z 5 O 100 Sei O 3 wn a7 75 2 D A M 5 5 E 50 TI E gt rt 25 0 50k 100k 150k 200k 250k 300k Room Volume in Cubic Meters Figure 25 Minimum distance to any measurement position from the excitation source e g a loudspeaker used for reverberation time measurements The minimum distance is a function of room volume estimated reverberation time and the speed of sound as described by Eq 1 above This example uses speed of sound at 20 C 68 F i e 343 6 meters sec or 1127 4 fps Directional Loudspeakers and Reverberation Time For the specific purposes of reverberation time measurement a potential complicating factor may arise if an installed sound system is to be used to excite the room Impulse response measurements made with directional loudspeakers typically have higher direct to reverberant ratios than IR measurements made by other methods which could affect
87. om with an installed sound system are you more interested in the room or the system Consider that using a directional loudspeaker to excite the space may affect reverberation times in locations that are on axis with the speaker Consider also that when using different speakers to measure from different points in the room any significant differences between those speakers will show up in your measurement results If the room is your target the course of least resistance might be to bring in an omnidirectional loudspeaker specifically designed for acoustical measurement On the other hand if your objective is to measure the performance of a loudspeaker system installed in a room you might be more concerned with early to late energy ratios and speech intelligibility metrics than reverberation time of the room exclusive of the sound system Direct vs Indirect IR measurement There are two basic ways to measure an impulse response in Smaart direct or indirect Or you could say there are three possible methods because the indirect IR measurement method that Smaart employs can be used as a deterministic or non deterministic measurement technique Let s start with the simplest and easiest to understand old fashioned direct IR measurement and work our way up from there This should not be construed as being in any sort of order of preference In fact we re starting with what is usually the least preferred method but they all have their selling p
88. onal Acoustics LLC All rights reserved to come in at lower and lower levels In theory this process goes on forever In practice the part we care about happens within a few seconds perhaps less than a second in smaller rooms and or spaces that have been acoustically treated to reduce their reverberation times The arrival of direct sound and probably some of the earliest arriving reflections will be clearly distinguishable on a time domain graph of the impulse response As reflected copies of the original sound keep arriving later and later at lower and lower amplitude levels they start to run together and form an exponential decay slope that typically looks like something close to a straight line when displayed on a graph with a logarithmic amplitude scale Anatomy of an acoustical impulse response Although no two non identical rooms ever have identical impulse responses there are a few component features that we can identify in some combination in almost any acoustical impulse response These include the arrival of direct sound early reflections reverberant build up and decay and the noise floor Figure 2 shows an acoustical impulse with its component parts labeled Descriptions for each follow 4p Log 18 ER Paf Arrival of Direct Sound 30 Early Decay 36 al Propagation Delay Reverberant Build up 42 4B 54 60 56 72 P y 550 650 7950 850 950 4 05k 1 15k 1 25k 1 35k 4 T Figure 2
89. or each available FFT DFT size Averaging and Overlap Averaging as we discussed earlier in this section is primarily something you concern yourself with when using effectively random stimulus signals With random or effectively random stimulus signals deciding 48 Page Copyright 2015 Rational Acoustics LLC All rights reserved how much is averaging is enough is kind of judgment call but typical settings are in the 4 16 range In very noisy environments you may want to use a larger value and or consider using a period matched signal When measuring with period matched noise or sweeps averaging is normally set to None or 2 although it is still possible that a higher setting could prove helpful if measuring in an extremely noisy environment Another factor that affects how averaging works is the Overlap setting found in Impulse Response options Options menu gt Impulse Response When overlap is set to 0 each FFT is calculated from unique data giving you the maximum amount of noise reduction that you can get from a given number of averages When you set the measurement Overlap to a non zero value then successive FFTs share some data in common see Figure 15 in the previous section remember that measurement overlap and Spectrograph overlap are two different things but the principle is the same If measurement overlap is set to 50 it only takes a little longer to record 16 averages than it would for 8 at 0 overlap You don t ge
90. ow level and gradually increase the gain until you reach the target excitation level or as close to it as you can reasonably get Input Levels Once you have your output levels nailed down adjust your input levels by whatever means available until both the measurement and reference signal levels labeled M and R on the control strip are roughly even and running at a reasonable level The yellow segment of Group DefaultTF L t the meters in Smaart runs from 12 dB to 6 dB full scale and that s the gege TF Pair TF Measurement w target zone The meters are peak reading and we hard limit peaks for E Cont noise signals in Smaart but you have to also allow for fluctuations due Ready to background noise in acoustical measurements and if you use noise from another source you may see wider variations in the peak levels With sinusoidal sweeps you can run the levels a little higher if you like due to the lower crest factor of the signal but you always want to keep Figure 26 The measurement signal level M is running a you will probably have to waste a few balloons or fire off a few blank comfortable level The reference the levels out of the red If you are doing a single channel measurement cartridges while adjusting the measurement channel gain to get a good R channel is clipping 47 Page Copyright 2015 Rational Acoustics LLC All rights reserved solid signal level with no clipping on the input level mete
91. peakers revealed reverb times as much as half a second shorter in the upper octaves than you might expect to see with an omnidirectional source In practice it may be possible to compensate for this difference by adjusting the upper end of the evaluation range to some point below the standard 5 dB start point but of course that represents a departure from standard practice and introduces an element of subjectivity that the standardized procedure was designed to eliminate Given a choice between a measurement made with a directional speaker or not making a measurement at all a less than ideal measurement is usually better than none However if any potential errors or subjectivity in evaluating reverberation time are a source of great concern for you it might be necessary to bring in an omnidirectional measurement speaker and do it by the book or at least record a few balloon pops just to have a second opinion Reverberation times aside it is worth mentioning that for most other purposes IR measurements made using an installed sound system that is actually used for amplified performances in the space you are measuring will be more representative of actual use of the system than measurements made by any other means In some cases in order to get everything you need you might need to make one set of measurements using an omnidirectional source positioned on the stage another using the installed sound system and perhaps even a third using the hous
92. r Measurement Duration If you are only concerned with measuring delay times then a good rule of thumb for how long the measurement needs to be is 3 times the longest delay time that you want to measure If you want to measure reverberation time and early to late energy ratios then the 60 dB decay time RT60 of the system is a good target This is kind of a functional requirement for period matched dual channel measurements but it s also a pretty good practical target regardless of how FFT 64k 1365 ms D you are measuring Ideally you would like to measure 30 dB of reverberant decay and the lower end of the evaluation range should be at least 10 dB ee Vote bi above the noise floor so that s 40 dB which of course is two thirds of 60 By the time you factor in propagation delay early decay and maybe enough of a noise tail to see the dynamic range of the measurement chances are you have FFT 64k 1365 MS eaten well into the remaining third Avg 512k 10922 ms 256k 5461 ms Of course both these rules require either knowing the delay time or Group 240k 5000 ms RT60 before you ve measured them That typically means you have to TF Pale 128k 2730 ms guess then measure then possibly adjust your guess and measure l 64k 1365 ms again For delay times you can use the distance to the source divided 32k 682 ms by the speed of sound as a Starting point For guesstimating 16k 341 ms purposes you can use 1130
93. r positions by hand you can save their positions to a comma separated values csv text file by selecting Save Decay Markers from the File menu To reload them again later first load the wav or aiff file containing the impulse response then select Load Decay Markers from the File menu to open the marker position file Reporting Results for Reverberation Time Reverberation time ideally should be measured from several locations throughout the room and the results from each measurement position averaged together octave band by octave band to get an average decay time for each octave Smaart doesn t do that part for you but the A Bands table does make it easy to get the data from each measurement into a spreadsheet Frequency Ranges The standard evaluation range for reverberation time is the six one octave bands from the 125 Hz 4 kHz Average times for each octave band can be presented in a table or on a graph When presenting reverb times on a graph the frequency axis of the graph should be labeled with the IEC standard nominal octave band center frequencies The y axis of the graph should have an origin of 0 and be labeled in seconds It should be noted both in the table and on the graph whether T20 or T30 was used ISO 3382 1 specifies that if a graph is presented it should be a line graph with a standardized aspect ratio of 2 5 cm per second and 1 5 cm per octave ISO 3382 2 isn t so picky It just says a graph Reverberation tim
94. rd Transform MLS The dual channel transfer function method that Smaart uses for indirect IR measurement also works best using period matched test signals but unlike the other three it can also produce very acceptable results using random test signals provided that both the reference and measurement signals are captured Transfer function based IR measurement systems work by calculating the frequency domain transfer function of a system under test SUT from the Fourier Transforms of two signals the signal going into a system and the output of the system in response to this input and then transforming the result back into the time domain using an inverse Fourier transform IFT Remember that perfect impulse that we were lamenting didn t exist in the real world for direct IR measurements Well that happens to be what you get if you take the IFT of the transfer function of two identical signals So it follows that when we take the transfer function of a stimulus signal and the SUT s response to it we theoretically should get something very much like its response to an ideal impulse 37 Page Copyright 2015 Rational Acoustics LLC All rights reserved And in fact that s pretty much what happens in practice when you use a period matched excitation signal When you use this same technique with effectively random signals you also get a lot of extra noise but repeating the measurement several times and averaging the results generally take
95. s care of Input Signal Output Signal that and Smaart makes this easy to do Discrete Measurement i Transfer Funct Fourier Signal Spectrum ransfer Function Transform DFT FFT Reference diese i ourier Signal Spectrum Impulse Response Transform d Dual Channel IR measurement Using Period Matched Signals In a way the discrete Fourier transform DFT or FFT all FFTs are DFTs but not all DFTs are fast is kind Figure 21 Block diagram of a Dual FFT transfer function IR measurement of a dirty mathematical trick Fourier transforms of all types theoretically work only with signals of infinite length but the DFT gets around this by pretending that a finite chunk of signal being analyzed is really just one instance of an infinitely repeating series of chunks that look exactly like it The best way to get around this inherent assumption of this cyclicality in DFT FFT analysis is to feed the DFT what it really wants to eat a test signal that either fits completely within the measurement time window or cycles with periodicity equal to the length of the DFT time constant Signals that meet these criteria can produce deterministic highly repeatable measurements in a fraction of the time it takes to get comparable results using random signals When using matched periodic signals e No data window is required 38 Page Copyright 2015 Rational Acoustics LLC All rights reserved e Delay compensation is not a critical r
96. signal at a given moment in time perhaps something like the one shown in Figure 13a 27 Page Copyright 2015 Rational Acoustics LLC All rights reserved An RTA is very useful tool but if you want to gain a better understanding how the spectrum of a signal changes over time you either need a really good memory or a different kind of graph One solution might be to just keep sliding the old data to the back instead of erasing it as new data comes in to form a 3 D graph with time on the z axis as in Figure 13b If you did this with a 3 D area chart instead of a bar graph it s commonly called a waterfall chart but let s continue with the bar graph example as both have the same limitation The problem with this approach is that as new data comes in higher level values in front will cover up some of the data in back so that you only get a partial picture as in Figure 13 a e b Magnitude dB Magnitude dB 63 125 250 500 1k 2k 4k 8k 16k Frequency Hz Magnitude dB Magnitude dB Figure 13 Turning a spectrum analyzer into a spectrograph You could rotate the graph in space until you can see all the bars Figure 12c but when you do that it becomes harder to discern how tall they all are Assuming that you have a color display waterfall charts were popular before anyone had color monitors you might try painting the tops of the bars different colors to better show their magnitudes as we
97. significant 40 Page Copyright 2015 Rational Acoustics LLC All rights reserved Dual Channel IR measurement Using Random Stimulus Signals An excitation signal that is not completely contained within or if continuous has its periodicity precisely matched to the time constant of a discrete Fourier transform being used for analysis is effectively random as far as the DFT is concerned In Smaart s signal generator the Random pink noise option or any pseudorandom cycle length with periodicity longer than the FFT size used are effectively random Periods shorter than the FFT size should never be used because they won t contain energy at all FFT bin frequencies Other examples would include music or noise signals with arbitrary periodicity from sources other than Smaart Perhaps the best argument in favor of using random signals for IR measurement is because you can If you want to make a measurement with music instead of noise you can If it s easier to generate pink noise from a mixing board or in a processor than it would be to inject a test signal into the signal chain from Smaart that will work The only absolute requirements are that the measurement system needs to capture an exact copy of the signal going into the SUT and that signal must contain enough energy at all frequencies of interest to you to make a solid measurement The main caveats associated with random stimulus signals are a relatively high level of noise meaning that you
98. stable however they shouldn t require very much adjusting under most circumstances Notice the five level marker widgets shown on the plot in Figure 9 If you were wondering about the cryptic labels your secret decoder and the default positions for each of the markers is as follows e Ld Level Direct This marker is positioned on the reverse integration curve at the point corresponding to the arrival time of direct sound e Le Level Early Decay This marker is automatically positioned 10 dB down from the Ld marker on the reverse integration curve The slope between Ld and Le is used to calculate EDT e Lr1 Level Reverberant 1 This marker designates the top of the reverberant decay range 5 dB down on the reverse integration curve from the Ld marker All of the level markers are user adjustable but positioning these three is pretty cut and dried You should rarely find any need to touch them e Lr2 Level Reverberant 2 This marker designates the end point for the reverberant decay slope If there is sufficient dynamic range it should be placed 30 dB down the reverse integration curve from Lr1 If not 20 dB will do Lr2 is one of the two markers that you may sometimes want to adjust by hand the other is Ln below e Ln Level of Noise This is typically the most subjective of the five markers in terms of placement The time location determines the start point for the reverse time integration curve which is the basis for positioning al
99. stem and a measurement signal post system This measurement type is essential for aligning loudspeaker systems as unlike RTA measurements TF measurements are not time blind This makes it possible to display Phase time Coherence stability and Magnitude response Power Amplification and Control Signal Generator Out Ref signal In Measurement Microphone Mic Meas Signal Computer USB or i 2 1 2 Firewire OUT IN Stereo Audio I O In this example Smaart s internal Signal Generator is used to excite the system The output of the generator is routed to input 1 of the I O Using a Y split the signal is physically looped back known as a hard wired loop back into an input channel on the I O This input is then assigned to the Reference signal of a transfer function measurement Multi channel I O In version 7 Smaart introduced the capability to run and view as many simultaneous real time Spectrum and Transfer Function measurement as well as live averages of individual measurements as your computer can handle Having multi channel l O capability enables you to set up multiple microphones to compare different measurement positions without a lot of running around This can be a huge advantage when working with larger more complex systems Traditional USB FireWire Stand Alone Interfaces Eight channel single rack space packages are common for multi channel audio interfaces The s
100. t Between 5 10 is Good 10 15 is rated as Fair and anything less than 15 is problematic 35 Page Copyright 2015 Rational Acoustics LLC All rights reserved 4 Measuring an Acoustical Impulse Response You can make an impulse response measurement in Smaart by clicking a single button Whether or not that measurement returns any useful information depends on decisions made beforehand The process for measuring an acoustical impulse response can be summarized as follows e Selection of measurement technique and stimulus type e Selection of excitation source s and position s e Selection of measurement position s e Estimation of the reverberation time and background noise e Selection of measurement parameters measurement length duration excitation level e Exciting the system and recording results What are we measuring and why Before you set out to make any acoustical measurement it is always helpful to define your objectives clearly The trip back to the site that you save may be your own Obviously we want to measure the acoustical impulse response of a system under test SUT for some reason but what exactly is the system Is it a room Is it a sound system Is it a combination of a sound system its acoustical environment What do you want to know about the system What equipment and measurements will be needed to make sure you get the information you need If you want to measure the reverberation time of a ro
101. t and resulting in unity gain summation Larger spacing would cause partial or full cancellation and combing Placing a microphone capsule on the center of a 61 Page Copyright 2015 Rational Acoustics LLC All rights reserved large surface can be useful when measuring for instance PA systems in an empty room devoid of seats as the reflective qualities of the floor will only be visible as a 6dB increase in overall level This method has to be used with caution though as the microphone will indicate mid high frequency sound pressure levels some 6dB higher than would be perceived at ear height in a full audience Audience members would of course still perceive a 6dB increase at low frequencies because ear height is acoustically close to the floor at long wavelengths but they d perceive a 6dB reduction at mid and high frequencies when compared to the microphone s empty floor response 62 Page Copyright 2015 Rational Acoustics LLC All rights reserved Appendix E Typical Measurement Rig Set Ups Below are some example Smaart measurement system setup diagrams for transfer function and dual channel IR measurement Basic connections are pretty much the same for any I O you may encounter We trust that you understand the audio cabling necessary connect your own equipment together Stereo 2x2 Audio I O Transfer Function or dual channel impulse response measurements are the result of comparing a reference signal pre sy
102. t sound the next most prominent features we tend to see are sound arriving the next most direct paths the lowest order reflections Sound that bounces off one surface to get from the excitation source to a measurement position is called a first order reflection two bounces gives you a second order reflection and so on Reflected sound can be useful or detrimental depending on factors such as its relative magnitude and timing in relation to the direct sound and the extent to which it is clearly distinguishable from the diffuse reverberant sound Early Decay Reverberant Build up and Reverberant Decay Following the arrival of direct sound and the lowest order reflections sound in a reverberant space will continue bouncing around a room for a while creating higher and higher order reflections At any given listening position some of this reflected energy will combine constructively over a relatively short period of time resulting in a build up of reverberant sound before air loss and absorption by the materials that make up reflecting surfaces begins to take their toll At that point the reverberant decay phase begins In practice you may or may not be able to see the reverberant build up in an impulse response as distinct from the direct sound and early reflections Sometimes it can be quite clearly visible other times not so much By convention the first 10 dB of decay after the arrival of direct sound in the reverse time integrated IR we wi
103. t the full benefit of averaging 16 unique FFTs in that case and processing time increases but you should see at least a little better signal to noise than you would get 8 with some net time savings Delay Compensation When making IR measurements with random signal sources you will get much better results if you compensate for the delay time through the system under test So plan on making the measurement twice if you don t already know the delay time once to find the delay and a second time for a keeper In Smaart 7 4 or higher the button labeled for Peak that appears next to the input level meters in IR mode see Figure 26 sets the reference signal delay to the highest peak in the impulse response Pushing the Button and Making the Measurement Having nailed down which measurement technique will get you the results you need chosen your excitation sources and measurement positions set your input and output levels and selected your FFT length and number of averages if applicable all that s left to do is push the button s For a dual channel measurement start your excitation signal unless you re using a triggered sweep in Smaart in which case it will start by itself and click the start button in the control strip on the right side of the main window see Figure 26 Smaart will take it from there and display the measurement results when it has finished For a single channel direct IR measurement click the record fi
104. t vertical pattern control can be very tightly controlled at mid and high frequencies And if aimed hard into the audience can reduce upper walls and ceiling excitation reducing the reverberant level and increasing the critical distance A note about echoes The upper part of a line array provides tight mid high pattern control above the array minimising direct ceiling excitation However the low radial attenuation rate can make them prone to create very audible echoes from distant boundaries if they aren t aimed accurately Always use Smaart s linear Lin IR graph to check for echoes as they can often be hidden amongst the rooms diffuse reverberation only to become audible especially on stage when the audience is in place and reverberant energy levels have dropped A simple way of understanding the line array effect of reducing radial attenuation from 6dB doubling of distance to perhaps 3 or 4dB doubling of distance at mid high frequencies is to imagine a listener very close to just one horn loaded element of the array In the figure above the listener let s call him Victor mainly hears element 5 with some minor off axis contributions from the other elements Each element 1 6 is a point source and would if measured in isolation have the usual 6dB doubling of distance radial attenuation characteristic However as Victor moves further away from element 5 he starts to benefit from the vector sum of elements 4 and
105. ted with early versions of Smaart A spectrograph of this file first appeared in the Smaart 1 0 user manual and it has popped up in innumerable places since then The measurement was recorded on the stage of a 6000 seat performance space using an overhead PA cluster as the excitation source It features a very prominent reverberant Figure 17 Spectrograph build up phase and problematic late reflected energy arriving about 160 ms dynamic range and color mapping after the direct sound This is a good file to experiment with to see how changing the FFT size Overlap and dynamic range can reveal different aspects of the IR You can see that we ve set the navigation pane graph type to ETC and moved Time 0 to about 100 ms The FFT size is 2K overlap is 95 and dynamic range is 20 to 60 dB 12 ETC V pam room wav 36 EE a Ue h BA SS aa U Sr i i gmi e 820 900 980 Calc 8k FFT 2k 5k x Overlap 94 800 500 300 200 T o Au T 80 50 30 20 80 140 200 260 320 380 440 500 560 620 680 740 800 860 920 980 31 Page Copyright 2015 Rational Acoustics LLC All rights reserved Figure 18 Broadband ETC and Spectrograph of a room impulse response showing a problematic back wall reflection The spectrograph can be extremely useful for examining both the level and frequency content of features in the IR such as reverberant build up and discrete reflections Speech Intelligibility Metrics We have discussed the appl
106. ters with respect to the wavelength of sound being produced Reading from left to right top to bottom you can either think in terms of an increasing diameter at a fixed frequency for each successive example or you can think in terms of increasing frequency for a fixed diameter for each successive example The nice thing about the directivity index figure DI is that it can be used in conjunction with a source s polar plot assuming a dB level scale to indicate the true directivity index for an off axis listener This is shown in the above top right example The on axis DI is 10dB but at the off axis point arrowed the polar response is 10dB down with respect to the on axis position The DI at the off axis point arrowed is the on axis DI minus the off axis figure i e the off axis DI is 10dB 10dB OdB Conventional loudspeaker arrays A spherical array of identical high Q loudspeaker horns will act like one large spherical radiator if the optimum inter cabinet splay angles are used Note however that the overall array directivity will tend to be lower than that of the individual horns at mid and high frequencies due to the array s wider overall coverage At lower frequencies however arraying can provide tighter pattern control by making the system acoustically larger Things can get quite complicated and pretty difficult to predict especially when you factor in manufacturing tolerances marketing optimized loudspeaker spe
107. tion and directional characteristics will determine both its ability to cover an audience consistently and its propensity to excite non audience areas of the room The latter can enhance the listening experience if for example it adds a nice clean tail of diffuse reverberation to an orchestral performance But room excitation can also ruin the auditory experience if for example loudspeaker lobes are causing coloration or if loudspeaker arrays are poorly aimed causing late reflections excessive reverberation and a loss of intelligibility Ignoring high frequency air absorption a point source s direct sound pressure will be inversely proportional to the source to listener distance i e 1 r or 6dB attenuation for every doubling of that distance In echoic rooms however reverberation will add to the direct sound at the listener position depending on how directional the source is An omnidirectional source away from any boundaries would radiate sound pressure spherically and cause strong room excitation Directivity factor Q If a sound source is directional so that its coverage is not fully spherical its sound power gets concentrated into a smaller sector of a sphere To help quantify that sound power concentration sound sources are Said to have a directivity factor Q Directivity factor Q is 1 for spherical source 2 for hemispherical source 4 for quarter spherical source and so on Sound pressure lev
108. typically aim for the lower end of 1 2 1 8 second range Spaces intended for symphonic performances and organ music can range from about 1 8 seconds up to three seconds or more in very large halls Reverberation times that are roughly equal across all frequencies are generally preferable for most purposes The exceptions are things like choral organ and romantic classical musical music where a reverberation time curve weighted more toward the lower frequencies may be preferred It s pretty normal for higher frequencies to decay faster than lows but you don t want to see times that are wildly different in neighboring octaves In general though acoustical treatments and or physical changes to the sound system are typically required to effectively address problems any problems you may find Early to Late Energy Ratios Early to late energy ratios are another way of objectively characterizing the reverberant characteristics of a room They are arguably a better measure than reverberation time for any venue where a sound system is an organic part of the acoustical equation They are simple to calculate automatically and are not subject to the kinds of complications that can make measurement of reverberation times somewhat subjective but they are a more recent innovation and may be less widely understood than RT60 Clarity Ratios C35 C50 C80 Clarity indexes are early to late energy ratios that compare the integral of the energy arriving with
109. ual channel IR measurement shares the specified percentage of data with the previous frame s e Show IR Peak sets the locked cursor in IR mode to the highest peak in the impulse response each time you run a new measurement e AlLCons Split Time sets the split time for the early to late energy ratio used in calculating ALCons a type of speech intelligibility estimation that can be calculated from an impulse response There is no real standard for this parameter but common settings are 10 or 20 milliseconds 12 Page Copyright 2015 Rational Acoustics LLC All rights reserved e Mag Threshold dBFS is similar to magnitude thresholding in transfer function measurements It is turned off when set to zero When set to any other value in dB FS Smaart will zero out the transfer function at any frequency where the reference signal does not cross threshold before calculating the dual channel impulse response Filter Settings Smaart includes a sweepable highpass filter for IR measurements that can be handy when analyzing IR measurements that include a lot of very low frequency noise or in cases where the reference signal being used in a dual channel measurement is band limited somehow The filter is applied post process to IR data for display purposes meaning it can be used for file based or newly measured data and only affects what you see on the screen It does not change the underlying measurement The cutoff frequency for the
110. uld measure two midrange drivers or other like devices and determine if they are wired with the same or different polarity by noting which direction the prominent peaks in the impulse are pointed Figure 4 shows a zoomed in view of the linear Lin scaled impulse response of a 2 order Butterworth bandpass filter with normal and inverse polarity Cutoff frequencies for the filter are 400 and 1600 Hz It s easy to see that the peaks in the two IRs are pointed in different directions relative to each other Unfortunately this doesn t necessarily tell you which one is correct But if you measured three like devices and one was different you might reasonably say that the majority rules Or if you 15 Page Copyright 2015 Rational Acoustics LLC All rights reserved measured two like devices and found opposite polarity and one of them sounded better it s possible you might have found the problem Linear view can also come in handy for looking at other types of signals in the time domain other than impulse responses Lin Y Lin V D H Figure 4 Zoomed in views of the linear impulse response of a bandpass filter with normal and inverse polarity One thing the Linear IR can tell you that the Log and ETC graphs can t is relative polarity For example you could measure two midrange drivers or other like devices and determine if they are wired with the same or different polarity by noting which direction the prominent peaks in the impulse
111. using at least two different stimulus source locations and reduces measurement uncertainty to no more than 2 5 for octave bands and 5 for 1 3 octave bands 45 Page Copyright 2015 Rational Acoustics LLC All rights reserved ISO 3382 2 specifies that all measurement positions should be at least one half wavelength apart and at least one quarter wavelength from any reflecting surface including the floor For example if we wanted to measure as low as the 125 Hz octave band the lower band edge is at 90 Hz At 68 F 20 C the speed of sound in air is 1127 4 feet per second 343 6 mps and so one wavelength at 90 Hz would be about 12 5 ft 3 8 m From that we could conclude that no two mic positions should be less than 6 25 ft 1 9 m apart and all microphones should be at least 3 13 ft 0 95 m above the floor and at least that far from any wall or other reflecting surface For the 63 Hz band you would need to double those distances Of course ISO 3382 1 applies specifically to measurement of reverberation time in rooms so what about acoustical measurements made for other purposes Two additional standards we could look at as a guide to microphone placement are ANSI S1 2 Criteria for Evaluating Room Noise and SMPTE 202M the current standard for calibrating cinema sound systems ANSI 1 2 has this to say about measurement positions Sound measurements for rating room noise under this standard shall be made at locations that are near
112. ution as a function of FFT size FFT sizes ranging from 512 samples to 4K samples are compared using 0 Overlap As the FFT size is increased frequency resolution improved but the peak of the IR is smeared out over a wider time range The x axis of the graph is time with frequency on the y axis Spectrograph Time and Frequency Resolution In Figure 13 we used an octave band analyzer as an example for the sake of simplicity Smaart 7 s spectrograph is narrowband but the concept is the same L Sat J Each vertical stripe in the IR mode spectrograph represents one FFT meaning the Etlan 0 FFT size determines both the time and frequency resolution of the display Larger 7 29 Page Copyright 2015 Rational Acoustics LLC All rights reserved FFTs provide greater detail on the frequency axis but may mask transient events on the time axis so it s a trade off in that respect The FFT and Overlap controls that appear below the Calc button determine the time Contiguous vs Overlapping FFTs and frequency resolution of the spectrograph display 0 Overlap Figure 14 illustrates how this relationship works It was created using the file 6dbOctlmpulse wav the impulse 50 Overlap response of a linear phase lowpass filter with 6 dB per octave roll off The sharpest part of the peak in the time gt impulse response where most of the HF energy lives is probably no more than a few milliseconds wide but Figure 15 With FFT overlap set

Smaart 7 Impulse Response Measurement and Analysis Guide

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