DESC9138 Assessment 1 Part 1: Question A: Briefly but carefully explain ways that could be used to check whether a room is anechoic. Explain what would be expected using the same techniques in a non-anechoic environment. Support your answer with reference to relevant articles/standards. Limit your answer to maximum 1A4 page.[3] An anechoic chamber can be defined as a room whose boundaries absorb all disturbance sound in a predefined frequency range, resulting in sound propagation equivalent to free-field conditions [1]. The qualification of an anechoic facility is dictated by the International Organization for Standardization, specifically, ISO 3745 and ISO 26101 [2]. These standards outline the minimum performance requirement for a facility to be classified as anechoic. To measure this, an approximated point source is moved along traverses from a sound source, and the sound pressure levels are recorded. The spatial decrease in sound pressure is then compared with those predicted by the inverse square law. The inverse square law can be described a 6dB reduction in sound pressure level per doubling of distance from a reference point. This is expressed as: (Week 3 Lecture) Once the measured sound has been compared with the inverse square law, the deviations can be compared with the tolerances defined by ISO 3745 [2]. These tolerances are divided into three one-third octave bands, whereby the ‘allowable difference’ ranges between 1 to 1.5 decibels [3]. When performance results fall outside of the predefined tolerances, the room is therefore deemed non-anechoic. This occurs due to the build-up of reflected sound contributing to an increase in sound pressure level relative to what is expected in free field sound environments. Question B: In the anechoic room tutorial (Week 3 module) sound was measured at three distances (0.75 m, 1.5 m, and 3 m) without and with the reflective ground plane. Analyse and explain the results in relation to theory. Comment on likely reasons for deviation from theory.[3] Measuring the sound from a single omnidirectional source in the anechoic room without a reflective plane highlighted some slight deviations from theory. While the inverse square law would suggest a 6-decibel reduction at each of the three distances, this was not strictly observed (see figure 1.) This is likely due to two key factors. Firstly, the increased sound reflections (reverberance) from people and equipment in the room, and secondly, equipment setup imperfections for example the precision of the measured distances. Figure 1. 0.75m 250Hz Change 1kHz Change 4kHz Change to -5.1dB -6.2dB -6.3dB 1.5m to 3m -6.6dB -5.4dB -6.2dB Given these baseline results, the addition of a wooden reflective plane meant that the inverse square law no longer applied. Instead of a 6dB reduction per doubling of distance, a hemispherical sound field in theory produces a 3dB reduction [4]. This theory however is not reflected across the results illustrated in Figure 2. Theory would suggest that this is a result of phasing, specifically, comb filtering. Comb filtering can be defined as an interference between a sound and a delayed version if itself. This out-of-phase relationship causes nulls and peaks in the received sound frequency response. The frequency of the first null is calculated 1 by multiplying the delay time by the period (f = 2π‘), the successive nulls occur at odd multiples while the peaks occur at even ones. Most notably, in the 250Hz band, a reduction of more than 8.1dB and 9.3dB suggests that the relationship between the sound signal, reflective plane and microphone was leading to considerable cancellations. According to (REFERENCE TEXTBOOK), lower frequencies are also more subject to phasing due to longer wavelengths. Additionally, the reflective ground plane (wood) has a greater absorption coefficient at lower frequencies, which could also be contributing to this result [5]. Figure 2. 0.75m to 1.5m -8.1dB -7.1dB -6.3dB 250Hz Change 1kHz Change 4kHz Change to 3m -9.3dB -4.3dB -5.5dB Question C: In the anechoic room tutorial (Week 3 module) sound was measured from two sources using a sound level meter halfway between them. Explain the theoretical expectations for the results and discuss the extent to which the measurement results meet expectations.[3] This exercise illustrates the effect of phase on sound pressure level through four setup variations. Firstly, the pink noise was played through each of the speakers producing a reading of around 75.1dB for speaker 1, and 75.3dB for speaker 2 (Figure 1.) These measurements provide the basis for further observations. Secondly, two coherent sounds were played through speakers 1 & 2. Theory would suggest that two inphase sounds of equal sound pressure level should result in a 6dB increase. This is illustrated by the equation: 75.1 75.3 75.1 + 75.3 = 20log10 (10^ 20 + 10^ 20 ) = 81.2dB 81.2 – 75.2 = 6dB Our test however resulted in a 5.6dB increase in sound pressure level (Figure 1.). This deviation from theory could be attributed to imperfections in the setup, whereby the geometry between the microphone to each of the speakers is not exactly equal, causing out-of-phase relationships to produce cancellations. Thirdly, one of the two channels was phase inverted 180 degrees (categorised by reversing the polarity of the signal). Theory would suggest that no sound would be recorded, however due to setup imperfections and interferences, this is highly unlikely, particularly in the high frequencies where wavelengths are extremely short and highly exposed to directivity discrepancies. Our testing resulted in a 4.2dB reduction (Figure 1.) Comb filtering can arguably be attributed to cancellations that decrease sound pressure level readings. And finally, two incoherent sounds were played simultaneously through speakers 1 & 2. According to theory, the addition of two incoherent sound pressure levels should produce an increase of 3dB (Figure 1.). Our results were consistent with these expectations, represented by the equation: 75.1 75.3 75.1 + 75.3 = 10log10 (10^ 10 + 10^ 10 ) = 78.2dB 78.2 – 75.2 = 3dB Figure 1. SOURCE SPEAKERS 1 & 2 COHERENT SOUND ONE CHANNEL PHASE INVERTED INCOHERENT SOUND DECIBEL READING 75.1dB & 75.3dB 80.8dB 71dB 78.2dB CHANGE (DB) +5.6dB -4.2dB +3dB Question D: In the anechoic room tutorial (Week 3 module) sound was recorded from two sources, with the microphone 2 m from one of the sources. Analyse the resulting sound recordings and discuss them in relation to theoretical expectations.[3] The analysis of pink noise recording revealed three key concepts, the inverse square law, interference (comb filtering) and directivity. Speakers 1 recorded a lower sound pressure level reading than speaker 2 due to the increased distance from source to receiver (see figure 1.) This is due to the dissipation of sound over distance, reflected by the inverse square law. According to theory, the difference in distance between sources 1 and 2 should yield a 1.36dB decrease in sound pressure level measured at speaker 1, represented by the equation: 2 20log10(2.34) = 1.4dB This was generally reflected in our testing, however with some exception. In the 8kHz to 16kHz octave-band the disparity between levels was considerably greater than expected (Figure 2.), which can arguably be attributed to speaker setup differences which resulted in beaming of the higher frequencies. This can be defined as a loudspeaker phenomenon whereby higher frequencies propagate with more directivity, resulting in greater fluctuations in sound pressure depending on where it is measured [6]. Figure 1. 1 2 2m Played through speakers 1 and 2 coherently, theory would suggest that destructive interference would be present, more specifically, a comb filter through the spectral effect of a delay between two identical sound sources. With a delay of 0.001s, calculated by dividing the distance difference by the speed of sound (0.34m/344m/s), we can calculate where the first peak (fundamental frequency) should theoretically be represented by: βπ = 1 0.001 = 1000Hz The subsequent peaks should therefore reside at integer multiples of the first peak (2000Hz, 300Hz, 4000Hz etc), and the nulls will occur at values equal to the peak frequency minus the original null frequency (500Hz, 1500Hz, 2500Hz etc) [7]. This, however, was not reflected in practice (figure 2.) arguably due to setup imperfections in the distances between sound sources and the receiver, along with the potential effect of unintended sound reflections (reverberation) from reflectors in the room (people & equipment). Figure 2. rec PinkNoise spk1 2 coherent 60 InChan1 40 20 Level [dB] 0 -20 -40 -60 -80 -100 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2 2.2 Frequency [Hz] 2.4 10 4 Incoherent noise played though speakers 1 and 2 could be expected to result in a 3dB increase in sound pressure level across the frequency spectrum the results largely matching this prediction (Figure 3.) Figure 3. DECIBELS (DB) Pink Noise 125Hz 250Hz Speaker 1 500Hz 1000Hz 2000Hz 4000Hz 8000Hz 16000Hz Speaker 2 Coherent Incoherent Utilising the same testing but with pure tones, coherent sounds played through speakers 1 and two produce results consistent with theory, whereby the peaks and nulls are at integers proportionate to the distance and delay time. This is illustrated in Figure 4 where the differences between single speaker sound levels and the coherent sound level at each of the frequencies was anticipated. This is because it is impossible for a pure tone to be incoherent, meaning they are less susceptible to the setup imperfections and the unintended reflections mentioned earlier [4]. Figure 4. DECIBELS (DB) Pure Tones 90 85 80 75 70 65 60 55 50 125Hz 500Hz 1000Hz 1500Hz 2000Hz FREQUENCY (HZ) Speaker 1 Speaker 2 Coherent Question E: Finally, in the anechoic room tutorial a reflective board was introduced with the source and microphone 0.61 m away from it. Analyse the resulting sound recordings and discuss how they relate to the two-source measurements.[3] This test produced results that are highly comparable to the previous exercise (Figure 1 & Figure 2.). This demonstrates that an image source created by a reflective plane is the equivalent to the addition of a second coherent sound source (given the geometric measurements are proportionate see Figure 1. Figure 1. 2m 1.22 1 2 Regarding pink noise, the slight differences in the spectral response seen in the 8kHz to 16kHz range (Figure 2.) could be caused by the porous nature of wood, whereby the viscous drag (friction) could cause a loss of momentum in air movements, resulting in lower sound pressure level readings at the source receiver [4]. Additionally, it could be speculated that the increase in sound level of the reflective plane source at 500Hz could be due to the resonant frequency of the wooden board. Also considered the natural frequency, this can increase oscillations at a given frequency resulting in higher amplitude [4]. Figure 2. DECIBELS (DB) Pink Noise 125Hz 250Hz 500Hz 1000Hz 2000Hz 4000Hz 8000Hz 16000Hz FREQUENCY (HZ) Two Coherent Speakers Reflective Plane Regarding the pure tones, there was negligible difference in results between the second speaker and the image source, which further illustrates the relationship (Figure 3.) Figure 3. DECIBELS (DB) Pure Tone 125Hz 500Hz 1000Hz 1500Hz FREQUENCY (HZ) Two Speakers Reflective Plane 2000Hz Part 2: a) Former students Guy Hopkins and Nicholas Lynar wrote the AARAE Analyser RoomNoiseEvaluation_NC_RNC, which is one of AARAE’s Level & Strength Analysers. This analyser is based on ANSI S12.2:2008, which provides methods to evaluate background noise in rooms using various methods including A-weighted sound pressure level, Noise Criterion (NC) and Room Noise Criterion (RNC). Use this to analyse the background noise recordings in the four auditoria. Comment on what the results reveal about the four auditoria. [/4] There are three concepts that will be discussed in relation to the background noise recordings provided: Aweighted sound pressure level, Noise Criterion and Room Noise Criterion. A-weighted sound pressure level (LA) is the average of maximum of sound pressure taken with a sound-level meter. It utilised a slow integration time and can be used to take a general measurement of noise created by air conditioning (HVAC) systems given no surging is present. Noise Criterion (NC) is used to quantify the appropriateness of noise levels for human occupancy. It does this by filtering the sound into octave band and averaging each band. The results can then be crossreferenced with a maximum acceptance level curve, derived from a survey. There are two rating methods within NC, tangency; based on the band level that intersects with the highest NC curve; and speech interference level (SIL); whereby the degree to which speech communication is disrupted by noise is quantified, done at the four octave bands of which human hearing is most sensitive to [8]. Room Noise Criterion (RNC) addresses the issue of surging in HVAC systems, which primarily concerned with the lower frequencies. This measurement method combines the 16Hz, 31.5Hz and 60Hz bands into one (critical band) and compares the maximum sound level and average sound level of this with the 125Hz band (fast integration time). The results are compared with a benchmark which indicates if large random fluctuations are present. If so, the RNC procedure adjusts the NC rating by placing more weighting on the frequency bands below 300Hz. The background noise recording for Wollongong Town Hall indicates that the noise level in the space is not optimal. While the LA conforms to the 30 to 35dBA recommendation outlined in Beranek’s article (31.3dBA), the rooms NC level of 20 (SIL) and 25 (tangency) falls outside the suggested range of 15 to 18. The data reveals that this is largely due to the 250Hz band ‘rumble’, seen in figure 1. Despite this however, surging has not been identified as an issue, therefore the RNC rating adjustment method is not necessary. Figure 1. According to the recording, Sydney City Recital Hall (AP) falls well within the recommended maximums for LA, NC and RNC. While the data might indicate a level in the excess of the recommendation, this has been influenced by the microphones internal system noise which dominates the 4kHz to 8kHz region and skews the readings. Taking this into consideration, it is clear that the hall presents no noise issues according to these testing parameters. Figure 2. Canberra’s Llewellyn Hall (LH) displays an adequate LA reading of 29.6dBA, however the NC both speech interference and tangency methods indicate higher than ideal ratings, 21 and 22 respectively. While the lower frequencies were not problematic (RNC rating adjustment was not necessary), it was the 2kHz band that greatly affected the evaluations, this is likely to be caused by internal noise caused by the measurement system. The Sydney Opera House Concert Hall (SOH) recording indicated that despite a 31.5dBA LA reading (withing the 30 to 35dbA recommendation), the NC analysis is well above the suggested level at 24 (SIL) and 25 (tangency). Unlike Wollongong Town Hall, this is due to higher frequency noise in the 1kHz to 2kHz range. Therefore, it can be suggested that the HVAC system needs optimisation [9]. b) Explain what is meant by the following terms and how they are calculated from a room impulse response: • early decay time • reverberation time T20 and T30 • clarity index C50 and C80 Developed by Wallace Clement Sabine in the 1890s, reverberation time (T) is the rate at which sound decays in room. Measured in seconds, it records how long it takes for a sound intensity level to drop 60dB [4]. To calculate this for a room, there a several steps. Firstly, we must find the energy loss in a room (πΌΜ ). This is done by: 1. finding the absorption of a surface (A), by multiplying the absorption coefficient of a medium (expressed as a ratio between 0 and 1) with its surface area (A = πΌπ). 2. calculating the rooms total absorption by summing the absorptions of all surfaces (A = ΣπΌπ) 3. dividing the rooms total absorption by its surface area, giving us an average absorption coefficient of ΣπΌπ a room (πΌΜ = Σπ ) Consequently, if the room has an average absorption coefficient greater than 0.15, it is prudent to calculate the room constant (R) for more accurate results in estimating the amount of sound in a room. This is done by multiplying the surface area and average room coefficient and dividing this by one minus the average Μ ππΌ absorption coefficient (R = Μ ). 1−πΌ After we have the average coefficient of a room (πΌΜ ), we then need to decide which formula to use to calculate reverberation time. If the average coefficient is less than 0.15, Sabine’s reverberation time formula is appropriate. To calculate this, there are three key steps: 1. Calculating the power absorbed by a room (PπΌ), expressed as soundfield energy (E) multiplied by speed of sound (c) multiplied by the surface area of a room (S) multiplied by the absorption coefficient (πΌ), πΈπππΌ divided by four ( 4 ). The number 4 is derived from the angle of incidence of a sound combined with its interaction with a surface, resulting in half a diffuse sound field (0.5), however given half the energy vectors are travelling away from the surface, this means that the diffuse field only interacts with 0.25 of a surface. 2. To extend on the power absorbed by a room, we need to consider the volume of a room multiplied by the ππΈ energy in the soundfield, divided by the decay constant (π). This is expressed as (PπΌ = π ). 3. Utilising the previous equations, we can now calculate reverberation time using Sabine’s formula. This is done by multiplying the decay constant (0.16, derived from the speed of sound relative to air temperature) 0.16π with the volume of a room (V), divided by the surface area of the room (A), expressed as π΄ . However, if the average absorption coefficient of a room greater than 0.15, Sabine’s formula can be inaccurate due to the way in which the formula assumes absorption occurs. In reality, absorption occurs at a fixed proportion on repeated reflection interactions with a surface, however Sabine’s formula treats the absorption as a single incidence. Therefore, given πΌΜ is high, it is practical to use the Norris-Eyring 0.16π Correction. This is expressed as T = −2.3ππππ Μ Μ Μ Μ 10(1− πΌ) Whereby the Sabine’s original formula is taken over a 60dB drop in sound (T60), this can present issues in the peak to noise ratio when the end of the reverberation decay meets the noise floor. To mitigate we can utilise early decay time benchmarks either T20 or T30 depending on how linear the reverse integrated decay curve is (or looking at to see if T20 or T30 provide consistent results). Early decay time (EDT) provides a simple estimation of the perceived reverberance. It is measured from the reverberation time decay curve between 0 dB to 10 dB below the original level. It is measured from the linear regression curve of an impulse response of which reverse integration has been applied. This is done by squaring and time-reversing the impulse response, cumulatively summing the wave and converting it back to decibels. T20 measures a -5 to -25dB decay time. While this is a less precise measurement, it can be utilised when the noise floor is high (less than -35dB), or in other words the recording quality is poor. T30 however implements a -5 to -35dB decay and requires a noise floor of less than -45dB, which can provide more accurate results as it is evaluated over a longer period provided the noise floor is less problematic. In either case the results are multiplied to give the reverberations time of T60 [10] [11]. Clarity Index is a way of measuring the subjective quality of speech transfer in a room. Expressed as a logarithmic ratio between early and late sound, it can be altered to optimise for speech or music clarity. C50 compares the first 50 milliseconds of sound energy from an impulse response with the subsequent sound energy (used for speech clarity), while C80 includes the first 80 milliseconds in the early sound, used for estimating music clarity. Another difference is the frequency ranges that are included in each. While C50 only incorporates the 2kHz band (most impactful on speech), C80 includes three frequency bands (500hz, 1000Hz and 1500Hz). They can be calculated by ten times the logarithm of the direct sound energy (Edirect) and early reflection energy (Eearly) divided by the late reflection energy (Elate). The time cutoff between Eearly and Elate is dictated by C50 or C80 [12] [2]. 100 - Finding Edirect sound is measured by π2 - Finding the Eearly is calculated by ( - Finding the Elate is calculated by ( 31200π π 31200π π )π −0.04π/π (1 - π −1.11/π ) )π −0.04π/π π −1.11/π) c) Calculate reverberation-related parameters using the AARAE analyser ReverberationTimeIR2 (use default settings of the analyser) from the room impulse responses. What do the results (especially those discussed in part b, above) tell us about each auditorium? How well do they relate to expectations from room volume and number of seats (check high quality research publications to justify your interpretation)? According to benchmarks outlines in Marshall Long’s ‘Architectural Acoustics’, Wollongong Town Hall (WH) exhibits acceptable acoustic properties [4]. With an average reverberation time of 1.5 seconds (T30) across the 125Hz-8kHz bands. This falls just below the recommended 1.8-2 second ‘ideal’ despite the limited seating to volume ratio. The early decay time is consistent with expectations (1.6 seconds), which suggests it is a reasonably ‘lively’ hall. C50 reading of -4dB places it inside the ideal range, suggesting the space has high speech intelligibility in the current setting [13]. Regarding clarity index, the C803 reading of 1.1dB falls well outside the recommended 0db to -4dB range, suggesting the room could be considered to be quite ‘dead’ for music purposes. It’s important to note that this index has been calculated using the 500Hz, 1000Hz and 1500Hz frequency bands, which is a correction put forth in Long’s book (Reference). Based on the ratio of seating to volume, it would seem that a high reverberation time would be present, however this is not the case, which suggests that the hall has had extensive acoustic treatment. Similarly, the Sydney Recital Hall (AP) displays a low reverberation time reading of 1.5 seconds (T30), representing the lower end of the recommended range (1.8-2 seconds). It is important to note that the 2000Hz frequency band was considerably influenced by the noise floor (indicated by a T20 to T30 ratio of 96%), and so T20 was used for this band. The early decay time averaged 1.6 seconds across the frequency bands, corresponding closely to the reverberation time. 1.8-2 seconds Clarity index C503 had a reading of 2.2dB while C803 was 0dB. These figures suggests that, similar to Wollongong Town Hall, the space is optimised for speech clarity over musical liveliness. A distinction between the two however is the seating to volume ratio. In Sydney’s Recital Hall the ratio is significantly higher, indicating that the seating may account for a large percentage of absorption occurring in the room (the chairs considered to have an absorption coefficient of 0.5-0.8) [14]. Llewellyn Hall in Canberra (LH) exhibits a longer reverberation time of 1.9 seconds (T30). This is expected given the significantly larger room volume than the previous two halls. An early decay time of 1.6 seconds suggests adequate liveliness. Clarity index C503 was -2.7dB while C803 was 0.25dB. This suggests the space is considerably ‘dead’ and optimised for speech clarity over musical envelopment. The low seating to volume ratio suggests that there are considerable amounts of acoustic treatment strategies in place as the absorption caused by seating would be contributing less than the other halls in this study. The impulse response of The Sydney Opera House Concert Hall (SOH) gives a reverberation time of 2.1 seconds (T30). This is longer than the 1.8-2 second recommendation outlines previously, suggesting the hall is optimised for musical liveliness as opposed to speech intelligibility. There was an early decay time of 2.1 seconds which further corroborates this idea. The clarity index C503 indicated a reading of -2.8dB and C803 was -1.75dB. According to this data, the hall may be boarding on ‘too reverberant’ despite an adequate C80 reading. This goes against expectations derived from the seating to volume ratio, which is second highest of the halls analysed. d) Listen to the room impulse responses using the various ways of playing sound provided by the main AARAE GUI. Identify the distinctive audible features of each impulse response. Comment on the extent to which the reverberation and clarity parameters correspond to audible differences between the impulse responses. Wollongong Town Hall (WH) This room sounds smaller than the others whilst also being more reflective, as though the space is dominated by hard surfaces (wood floors etc). The sound source seems slightly closer to the receiver than in the other tests given the lack of distinguishable early reflections. This auralization is partially consistent with the data. While it is true that the room is smaller, the clarity index would suggest it to be quite a ‘dead’ space, which was not my experience of the recording. The low seating to volume ratio is quite audible through the reverberation time of 1000Hz area. Sydney City Recital Hall Based on listening to the impulse response, the room sounds larger and with more soft materials than WH. There seems to be more low-mid reverberance, however this could be due to an increased distance from the sound source. It sounds moderately reflective with the presence of some early reflections. Consequently, I would assume the space would be better suited to musical envelopment (warmth) than WH. Llewellyn Hall, Canberra The impulse response of this room presents significantly more low frequency content over a longer reverberation time. This would have the effect of added warmth, contributing to a better sense of musical envelopment. The source sounds further away than in the previous two halls, and I would consider it to be a moderately reflective space. Comparing this assessment to the data, it is clear that C80 does not reflect my perception of its warmth. Sydney Opera House Concert Hall Most notably in this recording is the lack of low frequency content, which suggests the hall would lack warmth for musical envelopment. The hall sounds more reflective the other impulse responses, and the audible early reflections suggests the sound source was further away than in the previous. The reverberation time is distinctively longer than the other halls, with particular emphasis on the higher frequencies. This assessment reflects the data in that speech clarity is not optimal in this space. References [1] Russo M, KraljeviΔ L, Stella M, Sikora M. Acoustic performance analysis of anechoic chambers based on ISO 3745 and ISO 26101: standards comparison and performance analysis of the anechoic chamber at the University of Split [Internet]. Euronoise2018.eu. 2018 [cited 13 April 2021]. Available from: https://www.euronoise2018.eu/docs/papers/369_Euronoise2018.pdf [2] International Organization for Standardization. ISO 3382-1 Measurement of room acoustic parameters—Part 1: Performance spaces ISO, Geneva (2009) [cited 13 April 2021]. Available from: https://www.iso.org/standard/40979.html#:~:text=ISO%203382%2D1%3A2009%20specifies,and%20prese nting%20the%20test%20report. [3] Russo M, KraljeviΔ L, Stella M, Sikora M. 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Available from: https://support.biamp.com/General/Audio/Comb_filters#:~:text=The%20fundamental%20frequency%20of %20a,delay%20time)%2C%20in%20seconds [8] Errede S. Auditorium & Room Acoustics [Internet]. Courses.physics.illinois.edu. 2002 [cited 13 April 2021]. Available from: https://courses.physics.illinois.edu/phys406/sp2017/Lecture_Notes/P406POM_Lecture_Notes/P406POM_L ect9.pdf [9] Vér I, Beranek L. Noise and Vibration Control Engineering: Principles and Applications. 2nd ed. John Wiley & Sons, Inc.; 2005. [10] Gracey B. E - sound and vibration terms - acoustic glossary [Internet]. Acoustic-glossary.co.uk. 2021 [cited 13 April 2021]. Available from: https://www.acoustic-glossary.co.uk/definitionse.htm#:~:text=Early%20Decay%20Time%20(EDT)%20is,dB%20below%20the%20initial%20level. [11] Stauskis V. Changes in subjective acoustical indicators in halls with long and short reverberation times. Journal of Civil Engineering and Management [Internet]. 2004 [cited 13 April 2021];10(3):235-239. Available from: https://www.tandfonline.com/doi/pdf/10.1080/13923730.2004.9636311 [12] Cabrera D. Acoustic clarity and auditory room size perception [Internet]. Acoustics.asn.au. 2007 [cited 13 April 2021]. Available from: https://www.acoustics.asn.au/conference_proceedings/ICSV14/papers/p328.pdf [13] Ottley M. Designing for speech in a circular room [Internet]. Au.marshallday.com. 2018 [cited 13 April 2021]. Available from: https://au.marshallday.com/media/2802/speech-in-circular-rooms-aas2018-mott.pdf [14] Rossing T. Springer Handbook of Acoustics. Springer Science & Business Media; 2007.
