Presentation on theme: "The Relationship Between Audience Engagement and Our Ability to Perceive the Pitch, Timbre, Azimuth and Envelopment of Multiple Sources David Griesinger."— Presentation transcript:
The Relationship Between Audience Engagement and Our Ability to Perceive the Pitch, Timbre, Azimuth and Envelopment of Multiple Sources David Griesinger Consultant Cambridge MA USA
Overview This talk consists of an introduction, followed by sections from three fields. –The intro states the goal of the talk - to help build better concert halls and opera houses. –To do this we need to understand how acoustics affects the perception of sound. Part one – Physics – describes a physical mechanism by which human hearing may detect the pitch, timbre, azimuth and distance (near/far) of several sound sources at the same time, using frequencies the range of vocal formants (1000Hz to 4000Hz.) –The acuity of these perceptions is reduced in the presence of reflections and reverberation in a consistent and predictable fashion. –The consequence of this reduction in acuity is the perception of distance from the source, and a loss in excitement or engagement with the performance. –A computer model of this mechanism can be used to measure the psychological clarity of a hall from recordings of live music. Part two – Psychology – discusses the psychological importance of the perception of near and far on the ability of a sound to hold the attention of the audience. –And makes a plea for hall and opera designs that maximize audience engagement Part three – Acoustics – looks at the acoustic reasons certain concert halls are more engaging than others. –Hall shape does not scale. A shoebox shape that works for a hall with 2000 seats large hall will produce muddy sound over a wide range of seats if it is scaled to 1000 seats –Rectangular diffusing elements – coffers and niches – act as frequency dependent retro-reflectors, and are an essential ingredient in maintaining high clarity over a large number of seats.
Warning! Radical Concepts Ahead! The critical issue is the amount, the time delay, and the frequency content of early reflections relative to the direct sound. If the direct to reverberant ratio above 700Hz is above a critical threshold, early energy and late reverberation can enhance the listening experience. But, if not… Reflections in the time range of 10 to 100ms reduce clarity, envelopment, and engagement – whether lateral or not. –and the earliest reflections are the most problematic. Reflections off the back wall of a stage or shell decrease clarity –They are typically both early and strong – and interfere with the direct sound. Side-wall reflections are desirable in the front of a hall, but reduce engagement in the rear seats. –They are earlier, and stronger relative to the direct sound in the rear. Reflections above 700Hz directed into audience sections close to the sound sources have the effect of reducing the reflected energy in other areas of the hall – with beneficial results. –These features increase the direct/reverberant ratio in the rear seats –And attenuate the upper frequencies from side-wall reflections in the rear. –Coffers, niches, and/or open ceiling reflectors are invariably present in the best shoebox halls.
A few that work – note the rectangular coffers and niches Boston Symphony Hall Amsterdam Concertgebouw Vienna Grosse Musikverreinsaal Tanglewood Music Shed
Nice try … – and there are plenty more… Avery Fisher Hall, New York Alice Tully Hall, New York Kennedy Center Washington, DC Salle Playel, Paris
Introduction This talk is centered on the properties of sound that promote engagement – the focused attention of a listener. –Engagement is usually subconscious – and the study of its dependence on acoustics has been neglected in most acoustic research. –At some level the phenomenon is well known: Drama and film directors insist that performance venues be acoustically dry, with excellent speech clarity and intelligibility. As do producers and listeners of popular music, and customers of electronically reproduced music of all genres. The same acoustic properties that create excitement in a play or film can increase the impact of live classical music – but many current halls and opera houses are not acoustically engaging in a wide range of seats. –Halls with poor engagement decrease audiences for live classical music. Engagement is associated with sonic clarity – but currently there is no standard method to quantify the acoustic properties that promote it. –Acoustic measurement s such as Clarity 80 or C80, were developed to quantify intelligibility, not engagement. Venues often have adequate intelligibility – particularly for music – but poor engagement Acoustic engineers and architects cannot design better halls and opera houses without being able to specify and verify the properties they are looking for. –So we desperately need measures for the kind of clarity that leads to engagement.
The story of near, far, and harmonic coherence The author has been fascinated with engagement for a long time –particularly the perception of muddiness in a recording, and the lack of dramatic clarity in a hall or opera house. This fascination led to a discovery that a major determinant of engagement was the perception of near and far, – which humans can determine immediately on hearing a sound, even with only one ear, or with a single channel of recorded sound. –The perception has vital importance, as it subconsciously determines the amount of attention we will pay to a sound event. The importance of this perception, and the speed with which we make it, argue that determining near and far is a fundamental property of sound perception. –But how do we perceive it, and how can it be measured? In searching for the answer, the author found that engagement, near and far, pitch perception, timbre perception, and direction detection are all related to the same property of sound: –the phase coherence of harmonics in the vocal formant range, ~1000Hz to 4000Hz. Example: The syllables one to ten with four different degrees of phase coherence. The sound power and spectrum of each group is identical
Near, far, and sound localization The first step to the realization of the fundamental importance of phase coherence came from authors listening experience, which suggested that the perception of near and far is closely related to the ability to accurately identify the direction of a sound source. –When individual musicians in a small classical music ensemble sounded engaging and close to the listener, they could be accurately localized. –And when they sounded distant and non-engaging, they were difficult to localize. –Engagement is mostly sub-conscious and difficult to quantify – but localization experiments are relatively easy to perform – so I studied localization. Experiments with several subjects showed that the ability to localize sounds in a reverberant environment depends on frequencies between 700Hz and 4000Hz, –and that poor localization occurs when the sum of early reflections in the time range from 5ms to 100ms from any direction becomes stronger than the direct sound. –The earlier a reflection comes, the larger is its detrimental effect. With the help of localization data it was possible to construct a measure for the ability to localize sound in a reverberant environment. –The input to the measure is a measured or calculated binaural impulse response at a particular seat, – ideally with an occupied hall and stage.
Equation for Localizability – 700 to 4000Hz We can use a simple model to derive an equation that gives us a decibel value for the ease of perceiving the direction of direct sound. The input p(t) is the sound pressure of the source-side channel of a binaural impulse response. –We propose the threshold for localization is 0dB, and clear localization and engagement occur at a localizability value of +3dB. Where D is the window width (~ 0.1s), and S is a scale factor: Localizability (LOC) in dB = The scale factor S and the window width D interact to set the slope of the threshold as a function of added time delay. The values I have chosen (100ms and -20dB) fit my personal data. The extra factor of +1.5dB is added to match my personal thresholds. Further description of this equation is beyond the scope of this talk – but it is explained on the authors web-page.. S is the zero nerve firing line. It is 20dB below the maximum loudness. POS in the equation below means ignore the negative values for the sum of S and the cumulative log pressure.
Broadband Speech Data verifies the LOC equation: Blue – experimental thresholds for alternating speech with a 1 second reverb time. Red – the threshold predicted by the localization equation. Black – experimental thresholds for RT = 2seconds. Cyan – thresholds predicted by the localization equation.
Measures from live music Binaural impulse responses from occupied halls and stages are very difficult to obtain! –But if you can hear something, there must be a way to measure it. Part one of this talk describes a physiologically derived model of human hearing. The model arose from the search for a measure for near and far. –But the model is capable of explaining (and measuring) far more. –The model provides a means of separating sounds from multiple sources into independent neural streams, And allows independent analysis of each stream for pitch, timbre, and azimuth. The model may not be neurologically correct in detail –But it predicts many known properties of human hearing, and shows that all of them depend on the phase coherence of the incoming sound. –It provides a method that this phase coherence can be measured from binaural recordings of live music. This model is the subject of part one of this talk. Parts two and three show why the model is needed.
Part one – Physics Part one describes a physical mechanism by which human hearing could detect pitch, timbre, azimuth and distance (near/far) of several sound sources at the same time, using the phase coherence of harmonics in the range of vocal formants (1000Hz to 4000Hz.) –The model is built from functions that are known to be present in human hearing. Signals from the basilar membrane are analyzed not just for their average amplitude, but for modulations produced by interference between harmonics. –This information derives from the phase relationships between harmonics. A conceptually simple mechanism is suggested that allows the information from these modulations to be separated into independent neural streams, one for each sound source. –This this mechanism explains our uncanny abilities to detect the pitch, timbre, azimuth and distance of several sources at the same time. –And it also predicts the observed decrease in these abilities in the presence of reflections. The model need not be entirely correct to support the main point of this talk, which is that the models success in predicting what is and is not audible strongly supports the conclusions of part two and part three. –The phase coherence of harmonics in the vocal formant range give rise to our abilities to separate sounds from multiple sources, and independently perceive pitch, timbre, azimuth, and distance for each source. –The acuity of these perceptions is reduced in the presence of reflections and reverberation in a consistent predictable, and measureable fashion. –In the absence of this acuity sources become psychologically distant and non-engaging. The model provides a means for measuring the degree of phase coherence – and thus the engagement – in an individual seat using only live sound as an input.
Perplexing Phenomena The frequency selectivity of the basilar membrane is approximately 1/3 octave (~25% or 4 semitones), but musicians routinely hear pitch differences of a quarter of a semitone (~1.5%). –Clearly there are additional frequency selective mechanisms in the human ear. the fundamentals of musical instruments common in Western music lie between 60Hz and 800Hz, as do the fundamentals of human voices. –But the sensitivity of human hearing is greatest between 500Hz and 4000Hz, as can be seen from the IEC equal loudness curves. Blue: 80dB SPL ISO equal loudness curve. Red: 60dB equal loudness curve The peak sensitivity of the ear lies at about 3kHz. Why? Is it possible that important information lies in this frequency range?
More Perplexing Phenomena Analysis of frequencies above 2kHz would seem to be hindered by the maximum nerve firing rate of about 1kHz. –Why has evolution placed such emphasis on a frequency range that is difficult to analyze directly? A typical basilar membrane filter above 2kHz has three or more harmonics from each instrument within its bandwidth –How can we possibly separate them? How is it possible that in a good hall we can routinely detect the azimuth, pitch, and timbre of three or more sound sources (musicians) at the same time? –Even in a concert where a string quartet subtends an angle of +-5 degrees or less! (The ITDs and ILDs are miniscule…) Why do some concert halls prevent you from hearing several musical lines at once? –And what can be done about it? The hair cells in the basilar membrane respond mainly to negative pressure – they approximate half-wave rectifiers, which are strongly non-linear devices. How can we claim to hear distortion at levels below 0.1% ? Why do many creatures – certainly all mammals – communicate with sounds that have a defined pitch? –Is it possible that pitched sounds have special importance to the separation and analysis of sound?
Answers Answers become clear with two basic realizations: –1.The phase relationships of harmonics from a complex tone contain more information about the sound source than the fundamentals. –2. And these phase relationships are scrambled by early reflections. For example: my speaking voice has a fundamental of 125Hz. –The sound is created by pulses of air when the vocal chords open. –Which means that exactly once in a fundamental period all the harmonics are in phase. A typical basilar membrane filter at 2000Hz contains at least 4 of these harmonics. –The pressure on the membrane is a maximum when these harmonics are in phase, and reduces as they drift out of phase. –The result is a strong amplitude modulation in that band at the fundamental frequency of the source. When this strong modulation is absent, or noise-like, the sound is perceived as distant.
Basilar motion at 1600 and 2000Hz Top trace: A segment of the motion of the basilar membrane at 1600Hz when excited by the word two Bottom trace: The motion of a 2000Hz portion of the membrane with the same excitation. The modulation is different because there are more harmonics in this band. In both bands there is strong amplitude modulation of the carrier, and the modulation is largely synchronous. When we listen to these signals the fundamental is easily heard In this example the phases have been garbled
Nerve firing rates Nerve cells act like an AM radio detector, which recovers the frequency and amplitude of the modulation, while filtering away the frequency of the carrier. This picture shows the amplitude envelope of the previous picture, plotted in dB. It represents the rate of nerve firings from each band. The rate varies over a sound pressure range of 20dB. Like the detectors in AM radios, the hair cells (probably) include AGC (automatic gain control) – with about a 10ms time constant. The response over short times is linear, but appears logarithmic over longer periods.
AM Radio AM radio consists of a carrier at a fixed high frequency that has been linearly modulated by low frequency signals. An AM receiver half-wave rectifies the carrier, and filters out the high frequency components. –What remains is the recovered low frequency signals. –So an AM radio receiver uses a strongly non-linear device to recover a linear signal. But the rectification process can be viewed as a kind of sampling – it also produces aliases of the modulation. –In the case of an AM radio the aliases are at very high frequencies, and can be easily filtered away. –In the basilar membrane the carrier is close to the frequencies of the modulation – and the aliases can be problematic.
Amplitude modulation of a noisy carrier The motion of the basilar membrane when excited by phase-coherent harmonics appears to be an amplitude modulated carrier – but the carrier is not a fixed frequency, but an artifact of a filter with a finite bandwidth. –And the frequency of the carrier is within the audio band. Thus, rectification by the hair cells produces aliases that are both broad- band and highly audible. Spectrum of the syllable three from the rectified and filtered 2000Hz 1/3 octave band (blue) and the 2500Hz 1/3 octave band. (red) Note the fundamental frequency and its second harmonic are the same in both bands. The garbage is different.
Recovering a linear signal To recover a linear signal from these hair cells we need to have to combine and compare the outputs from many overlapping critical bands. –The aliases in each band are different because the carriers have different frequencies – but the modulations we wish to hear are nearly the same. Since for most signals the artifacts are not constant in time – we must also average the hair-cell firings over a period of time. –My data suggests an averaging time of 100ms. Because the carrier is broad-band, the aliases are also broad-band. – The signals are generally narrow band – so broad band signals may be ignored. Our hearing mechanism does all of these things.
An amplitude-modulation based basilar membrane model
A Pitch Detection Model A neural daisy-chain delays the output of the basilar membrane model by 22us for each step. Dendrites from summing neurons tap into the line at regular intervals, with one summing neuron for each fundamental frequency of interest. Two of these sums are shown – one for a period of 88us, and one for a period of 110us. Each sum produces an independent stream of nerve fluctuations, each identified by the fundamental pitch of the source.
Pitch acuity – A major triad in two inversions Solid line - Pitch detector output for a major triad – 200Hz, 250Hz, 300Hz Dotted line – Pitch detector output for the same major triad with the fifth lowered by an octave: 200Hz, 250Hz and 150Hz. Note the high degree of similarity, the strong signal at the root frequency, and the sub- harmonic at 100Hz
Summary of model We have used a physiological model of the basilar membrane to convert sound pressure into demodulated fluctuations in nerve firing rates for a large number of overlapping (critical) bands. Our physiological model of the frequency separation mechanism is capable of analyzing the modulations in each band into perhaps hundreds of frequency bins. –Strong, narrow-band signals at particular frequencies are selected for further processing The result: we have separated signals from a number of sources into separate neural streams, each containing the modulations received from that source. –These modulations can then be compared across bands to detect timbre, and IADs and ILDs can be found for each source to determine azimuth.
Advantages –The separated streams from each source can be easily analyzed for timbre, ITD and ILD with known neural circuits. –The model is conceptually simple – it is built out of a (large) number of building blocks that are known to exist in human neurology. It is easy to see how it could have evolved. –The circuit is fast. Useful data on timbre, ITD, and ILD is available within 20ms of the first input. As the sound is held pitch and azimuth acuity increases. –Because the ILD is created by high frequency harmonics, small differences in azimuth can create large differences in level Thus azimuth acuity is high enough to explain our ability to localize musicians.
Speech without reverberation: 1.6kHz-5kHz Note that the voiced pitches of each syllable are clearly seen. Since the frequencies are not constant, the peaks are broadened – but the frequency grid is 0.5%, so you can see that the discrimination is not shabby.
Speech with reverberation: RT=2s, D/R -10dB If we convolve speech with a binaural reverberation of 2 seconds RT, and a direct/reverberant ratio of -10dB the pitch discrimination is reduced – but still pretty good! The binaural audio sounds clear and close.
Speech with reverberation: RT=1s, D/R -10dB When we convolve with a reverberation of 1 seconds RT, and a D/R of -10dB the brief slides in pitch are no longer audible – although most of the pitches are still discernable, roughly half the pitch information is lost. This type of picture could be used as a measure for distance or engagement. The binaural audio sounds distant and muddy.
Two violins recorded binaurally, +-15 degrees azimuth Left ear - middle phrase Right ear - middle phrase Note the huge difference in the ILD of the two violins. Clearly the lower pitched violin is on the right, the higher on the left. Note also the very clear discrimination of pitch. The frequency grid is 0.5%
The violins in the left ear – 1s RT D/R -10dB When we add reverberation typical of a small hall the pitch acuity is reduced – and the pitches of the lower-pitched violin on the right are nearly gone. But there is still some discrimination for the higher-pitched violin on the left. Both violins sound muddy, and the timbre is poor!
Timbre – plotting modulations across critical bands Once sources have been separated by pitch, we can compare the modulation amplitudes at a particular frequency across each 1/3 octave band, from (perhaps) 500Hz to 5000Hz. The result is a map of the timbre of that particular note – that is, which groups of harmonics or formant bands are most prominent. This allows us to distinguish a violin from a viola, or an oboe from a clarinet. I modified my model to select the most prominent frequency in each 10ms time-slice, and map the amplitude in each 1/3 octave band for that frequency. The result is a timbre map as a function of time. –The mapping works well if there is only one sound source.
Timbre map of the syllables one two All bands show moderate to high modulation, and the differences in the modulation as a function of frequency identify the vowel. Note the difference between the o sound and the u sound.
Timbre map of the syllables one two with reverberation 2s RT -10dB D/R All bands still show moderate to high modulation, and the differences in the modulation still identify the vowel. The difference between the o sound and the u sound is less clear, but still distinguishable.
Timbre map of the syllables one two with reverberation 1s RT -10dB D/R The clarity of timbre is nearly gone. The reverberation has scrambled enough bands that it is becoming difficult (although still possible) to distinguish the vowels. A one-second reverberation time creates a greater sense of distance than a two second reverberation because more of the reflected energy falls inside the 100ms frequency detection window.
Non-coherent sources So far I have been considering only sources that emit complex tones with a distinct pitch. –What about sources that are not coherent, like a modern string section with lots of vibrato, or pink noise? Nearly any sound source – when band-limited – creates noise-like modulations in the filtered output. –Pink noise is no exception. Narrow-band filter it, and the amplitude fluctuates like crazy. Sources of this type cannot be separated by frequency into separate streams – but they can be sharply localized, both by ITD and ILD. This explains why in a good hall we can easily distinguish the average azimuth of a string section. If the strings play without vibrato they are perceived as a single instrument, with no apparent source width!
Example – Pink noise bursts with identical ILDs I created a signal that consists of a series of pink noise bursts, one of which is shown below. The noise is sharply high pass filtered at 2kHz. During the 10ms rise-time the noise is identical in the left and right channels. After 10ms, the noise in the right channel is delayed by 100us. The next burst in the series is identical, but the left and right channels are swapped. When you listen to this on headphones (or speakers) the sound localizes strongly left and right. Azimuth is determined by the ITDs of the modulations – not the onset
Summary of part 1: We have shown that the human ear has evolved to analyze fluctuations or modulations in the amplitude of the basilar membrane motion at frequencies above 1000Hz. –And not necessarily the average amplitude of the motion. –So long as the phases of the harmonics that create the modulations are not altered by reflections, the modulations from each source can be separated by frequency, and separately analyzed for pitch, timbre, azimuth, and distance. The modulations – especially when separated – carry more information about the sound sources than the fundamental frequencies. –And allow precise determination of pitch, timbre, and azimuth. All of these perceptions depend on the ears ability to perceive the direct sound from the source!!! (And there is currently no standard measure…) Reflections from any direction – particularly early reflections – scramble these modulations and create a sense of distance and disengagement. –But they are only detrimental to music if they are too early, and too strong. The C language model presented above makes it possible to visualize the degree to which timbre and pitch can be discerned in a recording of live music. –With calibration a single-number measure for engagement should be possible.
Direct sound and Envelopment Recent work by the author in both experiments with several subjects, and in live lecture demonstrations with loudspeakers, has shown that the sense of both reverberance and envelopment increases when the direct sound is separately perceived. –Where there is no perceivable direct sound the sound can be reverberant, but comes from the front. –When the direct sound is above the threshold of localization the reverberation becomes louder and more spacious. Envelopment and reverberance are created by late energy – at least 100ms after the direct sound. When the direct sound is inaudible the brain cannot perceive when a sound has started. –So effectively the time between the onset of the direct sound and the reverberation is reduced, and less reverberation is heard. –In the absence of direct sound syllabic sound sources (speech, woodwinds, brass, solo instruments of all kinds) are perceived as in front of the listener, even if reflections come from all around. The brain will not allow the perception of a singer (for example) to be perceived as all around the listener. In addition, Barron has shown that reverberation is always stronger in front of a hall than in the rear – so in most seats sound decays are perceived as frontal. –But when direct sound is separately perceived, the brain can create two separate sound streams, one for the direct sound (the foreground) and one for the reverberation (the background). A background sound stream is perceived as both louder and more enveloping than the reverberation in a single combined sound stream.
Time for Demos I will attempt to demonstrate the effects of the direct sound on the perception of distance, muddiness, and envelopment. The four speakers around the audience will play reverberation as it might exist in a hall. The center channel will produce the direct sound. The D/R will vary depending on where you are sitting. –And I will vary it to give everyone a chance to hear the effects near the threshold of audibility for the direct sound.
Part 2: Near, Far, and Engagement The apparent closeness of a sound source is a fundamental perception for all of us. –We can tell instantly if a person talking is within a few feet of us, or further away – and this perception has survival value. –The perception of Near depends critically on our ability to perceive the direct sound – the sound that travels to the listener without reflecting. –Surprisingly, in a theater or hall it is possible to perceive the performers as both acoustically close to the listener and enveloped by the hall. –The best halls (Boston Symphony Hall, Concertgebouw, the front half of the Musikverrein) provide both, but many, perhaps most, provide only reverberation. Harmonic coherence of speech and music is a principle cue for perceiving near and far. –The audio examples in the click box above show the decrease in apparent distance caused by increasing amounts of harmonic coherence. –Note that all of the examples have high intelligibility – but their emotional effect is quite different. The perception of near and far correlates with ability to localize sound sources – and the ability to separately hear several musical lines at the same time.
Amplitude modulation analysis – direct sound Spoken syllables one to ten analyzed by the neural model presented in Part one. Note the clear detection of the pitch of each vowel
Amplitude modulation analysis one to ten with 88ms all-pass reflections Note the pitch acuity has been reduced, along with the signal to noise ratio.
Amplitude modulation analysis one to ten with 133ms reflections The pitch discrimination with this analysis model (an early one) is poor with these reflections.
Experiences – Staatsoper Berlin Barenboim gave Albrecht Krieger and me 20 minutes to adjust the LARES system in the Staatsoper. My initial setting was much to strong for Barenboim. He wanted the singers to be absolutely clear, with the orchestra rich and full – a seemingly impossible task. Adding a filter to reduce the reverberant level above 500Hz by 6dB made the sound ideal for him. The house continues with this setting today for every opera. Ballet uses more of a concert hall setting – which sounds amazingly good. In this example the singers have high clarity and presence. The orchestra is rich.
Experiences – Bolshoi – a famously good hall for opera The Bolshoi is a large space with Lots of velvet. RT is under 1.2 seconds at 1000Hz, and the sound is very dry. Opera here has enormous dramatic intensity – the singers seem to be right in front of you – even in the back of the balconies. It is easy for them to overpower the orchestra This mono clip was recorded in the back of the second balcony. In this clip the orchestra plays the reverberation. The sound is rich and enveloping
New Bolshoi before modification The Semperoper was the primary model for the design of the new Bolshoi. As in Dresden the sound on the singers is distant and muddy, and the orchestra is too loud. RT ~1.3 seconds at 1000Hz. New Bolshoi Dresden What is it about the SOUND of this theater that makes the singers seem so far away? This theater suffers greatly from having the old Bolshoi next door! (The theater has received poor reviews from musicains and the press.)
Experiences – Amsterdam Muziektheater Peter Lockwood and I spent hours adjusting the reverberant level using a remote in the hall. –He taught me to hear the point where the direct sound becomes no longer perceptible, and the sonic distance dramatically increases. –With a 1/2 dB increase in reverberant level, the singer moved back 3-4 meters. –In Copenhagen, I once decreased the D/R by one dB while Michael Schonwandt was conducting a rehearsal. He immediately waved to me from the pit, and told me to put it back. Given a chance to listen A/B, these conductors choose dramatic intensity over reverberance. –When they do not have this chance, reverberation is seductive, and the singers be damned!
Experiences, Copenhagen New Stage We were asked to improve loudness and intelligibility of the actors in this venue. 64 Genelec 1029s surround the audience, driven by two line array microphones, and the LAREAS early delay system. A gate was used to remove reverberation from the inputs. 5 drama directors listened to a live performance of Chekhov with the system on/off every 10 minutes. The result was unanimous – it works, we dont like it. The system increases the distance between the actors and the audience. I would rather the audience did not hear the words than that this connection is compromised.
A slide from Asbjørn Krokstad - IoA,NAS Oslo 2008 To succeed: [in bringing new audience into concert halls…] ENGAGING Interesting "Nice [With permission] [We need to make the sonic impression of a concert engage the audience – not just the visual and social perceptions. Especially since audiences are increasingly accustomed to recordings!]
ENGANGEMENT, not NICE At the IoA conference in Oslo, Asbjørn Krokstad (a musician, conductor, and Norways best-known acoustician) gave a lecture where he insisted that acousticians needed to provide engagement, not just pleasant music. –And not just for drama and opera, but for chamber music and symphony too. –At the end of the lecture he showed a picture of the Teatro Colón in Buenos Aires, Argentina. Is this the concert hall of the future he asked? This hall is not a shoebox, but a large semicircular theater with a high ceiling. It ranks at the top in Beraneks surveys, and the reverberation time is 1.6 seconds occupied. Krokstad may have conducted there. Engagement requires the independent perception of the direct sound We must learn how to provide this essential element in halls. I have been fortunate to hear several of the live broadcasts of the Metropolitan Opera in a good theater. For example, the performance of Salome: –The sound was harsh and dry – radio mikes coupled to directional loudspeakers. But you could hear every syllable of Mattilas impeccable German. The performance was totally gripping! –This is the dramatic and sonic experience audiences increasingly demand.
Part 3 - Main Points The ability to hear the Direct Sound – as measured by LOC or through binaural recording analysis – is a vital component of the sound quality in a great hall. –The ability to separately perceive the direct sound when the D/R is less than 0dB requires time. When the d/r ratio is low there must be sufficient time between the arrival of the direct sound and the build-up of the reverberation if engagement is to be perceived. Hall shape does not scale –Our ability to perceive the direct sound – and thus localization, engagement, and envelopment - depends on the direct to reverberant ratio (d/r), and on the rate that reverberation builds up with time. –Both the direct to reverberant ratio (d/r) and the rate of build-up change as the hall size scales – but human hearing (and the properties of music) do not change. –A hall shape that provides good localization in a high percentage of 2000 seats may produce a much lower percentage of great seats if it is scaled to 1000 seats. –And a miniscule number of great seats if it is scaled to 500 seats.
Frequency-dependent diffusing elements are necessary, and they do not scale. The audibility of direct sound, and thus the perceptions of both localization and engagement, is frequency dependent. Frequencies above 700Hz are particularly important. –Frequency dependent diffusing elements can cause the D/R to vary with frequency in ways that improve direct sound audibility. –The best halls (Boston, Amsterdam, Vienna) all have ceiling and side wall elements with box shape and a depth of ~0.4m. These elements tend to send frequencies above 700Hz back toward the orchestra and the floor, where they are absorbed. (The absorption only occurs in occupied halls – so the effect will not show up in unoccupied measurements!) The result is a lower early and late reverberant level above 700Hz in the rear of the hall. This increases the D/R for the rear seats, and improves engagement. –The LOC equation is sensitive to all reflections in a 100ms window, which will include many second-order reflections, especially in small halls. Replacing these elements with smooth curves or with smaller size features does not achieve the same result. –Some evidence of this effect can be seen in RT and IACC80 measurements when the hall and stage are occupied. Measurements in Boston Symphony Hall (BSH) above 1000Hz show a clear double slope that is not visible at 500Hz. –The hall has high engagement in at least 70% of the seats.
Boston Symphony Hall, occupied hall and stage, stage middle to front of first balcony, 1000Hz Note the clear double-slope decay, with the first 12dB decaying at RT = 1s The direct sound is clearly dominant at this frequency in this seat. The sound is very good – Leo Beraneks favorite seat!
Boston Symphony Hall, occupied, stage to front of balcony, 250Hz But at 250Hz, there is no evidence of the direct sound. It has been swamped by reverberation.
We need better measures Current acoustic measures ignore both the D/R and the time gap between the direct (the first wavefront) and the reverberation. –RT, C80, and EDT all ignore the strength of the direct sound and the effects of musical style on the audibility of the D/R. –IACC comes close, but measures only lateral reflections. LOC and my hearing model attempt to supply a simple measure for a basic human perception which depends on direct sound. –Impulse response measurements under occupied conditions are notoriously difficult to obtain. –But this problem can be circumvented through binaural recordings of live sound. The hearing model presented in part one promises to supply measures that use binaural recordings of actual performances as inputs. –And the ability to listen to these recordings to test the validity of these measures against the true experience.
Why do large halls sound different? In Boston Symphony Hall (BSH), and the Amsterdam Concertgebouw (CG) the reverberation decay is nearly identical, but the halls sound different. –The difference can be explained using the same model that was used to develop LOC. –Lacking good data with an occupied hall and stage I used a binaural image-source model with HRTFs measured from my own eardrums.
Reverberation build-up and decay – from models Amsterdam Boston The seat position in the model has been chosen so that the D/R is -10dB for a continuous note. The upward dashed curve shows the exponential rise of reverberant energy from a source with rapid onset and at least 200ms length. The reverberation for the dotted line is exponentially decaying noise with no time gap. The solid line shows the image-source build up and decay from a short note of 100ms duration. Note the actual D/R for the short note is only about -6dB. The initial time gap is less in Boston than Amsterdam, but after about 50ms the curves are nearly identical. (Without the direct sound they sound identical.) Both halls show a high value of LOC, but the value in Amsterdam is significantly higher – and the sound is clearer. LOC = +6dB LOC = 4.2dB
Comparisons of C80, C50, IACC80, and LOC Conventional measures for the models of Amsterdam Concertgebouw and Boston Symphony Hall give the following results: Amsterdam: C80 =.43dB, C50 = -2.8dB, IACC80 =.38, LOC = +6dB BSH: C80 =.65dB, C50 = -2.1dB, IACC80 =.22, LOC = +4.2dB Half-Size BSH: C80 = 3.7, C50 = 1.7, IACC80 =.15, LOC = 0.5dB Only the IACC80 shows that Amsterdam might have more direct sound than Boston. The standard Clarity measures predict the opposite – and predict that the small hall would have high clarity, and it does not. But IACC80 is sensitive only to lateral reflections. Strong reflections from the front, overhead, or rear do not affect IACC. An IACC of 0.22 would usually be considered too low for good sound. In spite of this BSH has both clarity and good localization in this seat.
Smaller halls What if we build a hall with the shape of BSH, but half the size? –The new hall will hold about 600 seats. –The RT will be half, or about 1 second. –We would expect the average D/R to be the same. Is it? How does the new hall sound? –If the client specifies a 1.7s RT will this make the new hall better, or worse?
Half-Size Boston The gap between the direct and the reverberation and the RT have become half as long. Additionally, in spite of the shorter RT, the D/R has decreased from about -6 in the large BSH model, to about -8.5 in the half-size model. This is because the reverberation builds-up quicker and stronger in the smaller hall. The direct sound, which was distinct in more than 50% of the seats in the large hall will be audible in fewer than 30% of the seats in the small hall. If the client insists on increasing the RT by reducing absorption, the D/R will be further reduced, unless the hall shape is changed to increase the cubic volume. The client and the architects expect the new hall to sound like BSH – but they, and the audience, will be disappointed. As Leo Beranek said about the Berlin Philharmonie: They can always sell the bad seats to tourists. LOC =0.5
Great Small Halls Exist! Jordan Hall at New England Conservatory has 1200 seats, an RT of 1.3s fully occupied. The shape is half-octagonal, with a high ceiling. The audience surrounds the stage, with a single high balcony. The average seating distance is much shorter than a shoebox hall, increasing the direct sound. The high internal volume allows a longer RT with low reverberant level. The sound in nearly every seat is clear and direct, with a marvelous surrounding reverberation. But the stage house is deep and reverberant. Small groups always play far forward. Although the hall is renowned as a chamber music hall, it is also good for small orchestras and choral performances. It was built around The hall is in constant use – with concerts nearly every night, (and many afternoons.)
Williams Hall, NEC Williams hall, in the same building, has ~350 seats in a square plan with a high ceiling. The sound from a piano sound is clear and reverberant in most, if not all, seats. (The audience usually sits where the orchestra is rehearsing in this picture.) The square plan keeps the average seating distance low. The high ceiling and high single balcony provides a long RT without a high reverberant level. The absorbent stage eliminates strong reflections from the back wall. By absorbing at least half the backward energy from the musicians, the stage increases the d/r. Note the coffered ceiling – similar to BSH.
Hard learned lessons Where clarity is a problem in small halls, acousticians usually recommend adding early reflections – through a stage shell, side reflectors, etc. –We tried this in a small hall by placing plywood panels behind the piano. The sound became louder and less clear. Just the opposite of what was needed. These measures reduce the gap between the direct sound and the reflected energy and decrease LOC. –They increase loudness – which is usually already too high, while increasing the sense of distance to the performers. –A better solution is to add absorption, or perhaps some means of deflecting the earliest reflections to the ceiling, or into the front of the audience where they can be absorbed. Re-direction tricks of this nature do not work well in small halls, as the second and third order reflections they create will arrive within the 100ms window that determines LOC. –Small halls have strong direct sound and too many early reflections The early reflections also come too quickly. Adding more reflections is exactly the wrong thing to do. –Adding absorption will improve clarity but reduce the late reverberant level and the RT. Electronics, or more cubic volume, can restore the longer RT without decreasing the D/R. In practice, not everyone is aware of, or appreciates, engagement. It is mostly a subconscious perception. Reverberation or resonance is immediately apparent to everyone – which is why it has become so over-emphasized in hall design. –Adding absorption may not be appreciated by everyone unless the decrease in late reverberation can be compensated. –Such compensation can be surprisingly easy. Adding a few tenths of a second to the late reverberation time of a small hall can be accomplished electronically with very few loudspeakers. The result can be beautiful and completely transparent.
In the best halls the reverberant level is lower than would be expected from classical acoustics D/R is frequency dependent in halls, and frequencies above 700Hz are particularly important for engagement. –Surface features can be used to decrease the reflected energy level in the rear of the hall at higher frequencies. In addition, the distribution of absorption in a hall significantly alters the distribution of the reflected energy. –In a good hall absorption is highly non-uniform. A high ceiling with a lot of reflecting surfaces above the audience can increase RT without increasing the reflected energy level near the audience. The reverberation created tends to stay up near the ceiling. –This helps to keep the D/R above ~700Hz constant over a large number of seats. –Current modeling techniques may not properly calculate these effects. Old fashioned light models might work better…
Hall Shapes and direct-sound perception threshold as a function of size A large hall like Boston has many seats above threshold, and many that are near threshold If this hall is reduced in size while preserving the shape, many seats are below threshold It is better to use a design that reduces the average seating distance, using a high ceiling to increase volume. Boston is blessed with two 1200 seat halls with the third shape, Jordan Hall at New England Conservatory, and Sanders Theater at Harvard. The sound for chamber music and small orchestras is fantastic. RT ~ 1.4 to 1.5 seconds. Clarity is very high – you can hear every note – and envelopment is good. Above threshold Near threshold Below threshold
Retro reflectors above 1000Hz Boston, Amsterdam, and Vienna all have side-wall and ceiling elements that reflect frequencies above 1000Hz back to the stage and to the audience close to the stage. This sound is absorbed – reducing the reverberant level in the rear of the hall without changing the RT. Another classic example is the orchestra shell at the Tanglewood Music Festival Shed, designed by Russell Johnson and Leo Beranek. Many modern halls lack these useful features!!!
High frequency retro reflectors Rectangular wall features scatter in three dimensions – visualize these with the underside of the first and second balconies. High frequencies are reflected back to the stage and to the audience in the front of the hall. The direct sound is strong there. These reflections are not easily audible, but they contribute to orchestral blend. But this energy is absorbed, and thus REMOVED from the late reverberation – which improves clarity for seats in the back of the hall. Examples: Amsterdam, Boston, Vienna
High frequency overhead filters A canopy made of partly open surfaces becomes a high frequency filter. Low frequencies pass through, exciting the full volume of the hall. High frequencies are reflected down into the orchestra and the audience, where they are absorbed. Examples: Tanglewood Music Shed, Davies Hall San Francisco In my experience (and Beraneks) these panels improve Tanglewood enormously. They reduce the HF reverberant level in the back of the hall, improving clarity. The sound is amazingly good, in spite of RT ~ 3s. In Davies Hall the panels make the sound in the dress circle and balcony both clear and reverberant at the same time. Very fine… (But the sound in the stalls can be loud and harsh.)
The necessity of occupied measurements The effects of frequency dependent reflecting elements depends on the presence of absorption on the stage and the front of the audience. Measuring the halls without absorption in these areas will not detect these vital effects. In addition, engagement is highly dependent on the D/R ratio – and this is also not correctly measured in an unoccupied hall. Thus measurement of localization and engagement requires that both hall and stage be occupied!
Binaural Measures The author has been recording performances binaurally for years. Current technology uses probe microphones at the eardrums. We can use these recordings to make objective measurements of halls and operas. The hearing model described in part one can be used to measure the phase coherence in these recordings.
Some demos of eardrum recordings These recordings have been equalized for loudspeaker reproduction. You may be able to judge clarity and intelligibility over near-field loudspeakers. –Accurate headphone reproduction requires headphone equalization –A method of equalizing headphones through equal loudness measurements is described in another paper on the authors web-site. –In general large circumaural headphones do not work well even when equalized. On-ear phones, such as the Sennheiser 250 or 100, can work better. opera balcony 2, seat 11 –Moderate intelligibility, reverberant sound. –OK for non-Italian speakers with subtitles opera balcony 3, seat 12 –Poor intelligibility, very reverberant opera standing room –Deep under balcony 2 – good intelligibility –This was preferred by Italian speakers A concert hall – row 8 (quite close) –Very good sound. Not so good further back.
Conclusions Performance venues should maximize engagement over a wide range of seats, while at the same time providing adequate late reverberation. To achieve this goal the direct sound must be perceived by the brain as distinct from the reflected energy – and this includes early reflections from all directions. –Engagement is a spatial property that is sensitive to medial reflections. It can be heard with only one ear (and measured with one microphone). –But it is essential to measure with both hall and stage OCCUPIED! The perception of reverberance and envelopment also depends on the audible presence of direct sound. –In the presence of adequate late reverberation direct sound increases envelopment and reverberation loudness. The audibility of direct sound depends on the D/R ratio above 700Hz, and the time delay of reflections in the first 100ms. –Hall sound can often be improved by frequency dependent reflecting elements, or by adding absorption to the stage rear wall. The optimum value for the D/R ratio depends on the hall size – –The D/R ratio must increase as hall size is reduced if clarity, localization, and the sense of envelopment is to be maintained. –D/R and engagement can be increased by decreasing the average seating distance, decreasing the reverberation time, increasing the hall volume, or by careful use of rectangular diffusing elements. –This is particularly true in opera houses and halls designed for chamber music. –A 1.8 second reverberation time is NOT necessarily ideal in a 1000 seat hall!!! Remember that changes in reverberant LEVEL (D/R) and initial time delay are more audible than changes in RT. To maintain clarity, low sonic distance, azimuth detection and envelopment in a small hall (and many large halls) it is desirable to reduce the average seating distance, and widely diffuse or absorb the earliest reflections, whether lateral or not. –The best small halls do this already. Current hall measurements ignore both the D/R and the time delay between direct sound and reverberation. This talk introduces methods to overcome this lack.
Appendix Slides cut from the original presentation. They may – or may not, be helpful to understanding some of the concepts.
Number of taps In our model the number of taps for each frequency is simply the number of taps that fits in the delay line given the period we wish to measure. –If W is the length of the delay line in seconds, and f is the frequency in Hz, then N, the number of taps is: N(f) = W*f In our model the output of each summing neuron is divided by N, so all frequencies have roughly the same amplitude. In a comb filter of this type, the frequency sharpness of each summing node (full width at half maximum - fwhm) is just the frequency divided by N. fwhm = f/N In practice the sharpness is higher, as the input to the filter is not sinusoidal, but sharply peaked at the frequency of the fundamental. Our model achieves a frequency discrimination (+-3dB) of 1%.
Detection of harmonics and sub-harmonics Comb filters are sensitive to harmonics of the tap period. –The full width at half maximum is sharper for the harmonic than for the fundamental. –Thus a modulation that contains harmonics will be more accurately detected. A tap sequence intended to detect 100Hz will also detect 200Hz and 300Hz. –This property produces what appear to be sub-harmonics of the input modulations, since a 200Hz modulation will also excite the 100Hz tap sequence. This artifact has interesting consequences for harmony, as the pitch pattern produced by both major and minor triads has a strong output at the root frequency –Regardless of the inversion of the notes in the triad.
Minor triad in two inversions Solid line: output of the pitch detector with a minor triad, 200Hz, 235Hz, and 300Hz. Dotted line: The same triad with the fifth lowered by one octave. 200Hz, 235Hz, and 150Hz. Notice the acuity of the pitch detection is better than 1%, or 1/6 th of a semitione. Each peak represents a separate neural data stream, which can be further analyzed for timbre and azimuth.
Parameters The model contains several parameters: 1. The choice of a log-linear model for the hair cells, not a fully logarithmic detector. (the choice is clear…) – 1600,2000Hz bands after hair cell: Source - Logarithmic Log-linear 2. The choice of 10ms as the response time of the logarithmic adaptation. –This causes the onsets of sounds to have a higher amplitude than the body of the sound, which is probably an advantage 3. The choice of a 100ms window for the frequency filter –This choice is consistent with earlier work on localization. 4. The use of equal weighting for all the taps –Chosen for simplicity. Further work is needed. But equally weighted taps work rather well…
Why pitch is robust in the presence of reverberation Although pitch acuity is reduced by reverberation, reverberation is not capable of completely scrambling the phases of the harmonics. Some 1/3 octave bands will still contain significant modulation – but which have modulation, and which do not, is random, and varies from note to note. If we are relying on the amplitude of the modulations in each band to determine timbre – then timbre will be scrambled. In fact the major perception of a distant source is that the timbre has become muddy.
What is Auditory Engagement Engagement is the perception that you are not just watching a scene from distance, but present in the middle of it. –Thus lack of distance is a critical component of presence. Auditory engagement is the perception that you are acoustically close to the sound sources. –Distance is perceived directly through harmonic coherence – but experiments to directly measure it with subjects are difficult. However it correlates both with the ability to localize sound sources, and the perception presence, or musical clarity. –To perceive presence you must be able to localize sound sources nearly all the time, –and be able to distinguish them from one another nearly all the time. Clear localization and the ability to hear most of the notes are key components of audience engagement. –Although particularly important in drama and opera, it should be (and often is not) a part of the emotional experience of music. –Being able to hear all the notes and localize the players draws the audience into the performance. They dont just watch it. This view of clarity is different from the one that equates clarity with intelligibility. Perhaps we need a new word for it.
Experiment for threshold of Azimuth Detection in halls A model is constructed with a source position on the left, and another source on the right Source signal alternates between the left and a right position. When the d/r is less than about minus 13dB both sources are perceived in the middle. Subject varies the d/r, and reports the value of d/r that separates the two sources by half the actual angle. This is the threshold value for azimuth detection for this model (Above this threshold the subject also reports a decrease in subjective distance)
Threshold for azimuth detection as a function of frequency and initial delay As the time gap between the direct sound and the reverberation increases, the threshold for azimuth detection goes down. (the d/r scale on this old slide is arbitrary) As the time gap between notes increases (allowing reverberation to decay) the threshold goes down. To duplicate the actual perception in small halls I need a 50ms gap between notes.
An important caveat! All these thresholds were measured without visual cues The author has found that in a concert (with occasional visual input) instruments (such as a string quartet) are perceived as clearly localized and spread. When I record the sound with probes at my own eardrums, and play it back through calibrated earphones the sound seems highly accurate, but localization often disappears! –Without visual cues when the d/r is below threshold the individual instruments are localized and spread when they play solo, but collapse to the center when they play together. –My brain will not allow me to detect this collapse when I am in the concert hall – even if I close my eyes most of the time! –With eyes closed it is more difficult to separate the sounds of the individuals, such as the second violin and the viola. This difficulty persists in the binaural recording.
Localization For this paper we assume sound sources are localized by the direct sound. –In some cases localization is aided by early reflections – but these vary strongly from seat to seat, and are too complex to consider here. For localization to be successful the direct sound must be perceived. –Prompt strong reflections can – and do – mask the direct sound. Lets propose that the brain detects the loudness of – and the presence of – sounds by integrating nerve firings over a period of time. –If the integrated nerve firings from the direct sound exceed the integrated nerve firings from the reflections inside this time window, the direct sound will be perceived – and localized. We can calculate the threshold of perception by double integrating the impulse response over a fixed time window.
The ear perceives notes – not the impulse response itself. Here is a graph of the ipselateral binaural impulse response from spatially diffuse exponentially decaying white noise with an onset time of 5ms and an RT of 1 second. This is NOT a note, and NOT what the ear hears! To visualize what the ear hears, we must convolve this with a sound. –Lets use a 200ms constant level as an example. The nerve firings from the direct component of this note have a constant rate for the duration of the sound. The nerve firings from the reverberant component steadily build up until the note ceases and then slowly stop as the sound decays. D/R = -10dB RT = 2s: C80 = 3.5dB C50 = 2.2dB IACC80 =.24 RT = 1s: C80 = 6.4dB C50 = 4.1dB IACC80 =.20
Direct and reverberation for d/r = -10dB, and RT = 1s The blue line shows the rate of nerve firing rate for a constant direct sound 10dB less than the total reverberation energy. The red line shows the rate of nerve firings for the reverberation, which builds up for the duration of the note. The black line shows a time window (100ms) over which to integrate the two rates. In this example the area in light blue is larger than the area in pink, so the direct sound is inaudible.
Direct and build-up RT = 2s If we hold the d/r constant, when the reverberation time is two seconds it takes longer for the reverberation to build up, so the light blue area decreases, while the pink area stays constant. This makes the direct sound more audible. In a large hall the time delay between the direct sound and the reverberation also increases, further reducing the area in light blue. The direct sound would be even more audible.
Equation for Localizability – 700 to 4000Hz We can use this simple model to derive an equation that expresses the ease of perceiving the direction of direct sound as a decibel value. p(t) is the sound pressure of the ipselateral channel of a binaural impulse response. With the previous simple assumptions, we propose the threshold for detection would be 0dB, and clear localization would occur at a localizability value of +3dB. Where D is the window width (~ 0.1s), and S is a scale factor: Localizability (LOC) in dB = The scale factor S and the window width D interact to set the slope of the threshold as a function of added time delay. The values I have chosen (100ms and -20dB) fit my personal data. The extra factor of +1.5dB is added to match my personal thresholds. S is the zero nerve firing line in the previous two slides. It is 20dB below the maximum loudness. POS means ignore the negative values for the sum of S and the cumulative log pressure.
Some explanation of the equation The equation as written in the previous slide simply calculates the ratios of the pink and blue areas shown in the previous pictures. The first integral on the left in LOC is the pink area – the sum of the nerve firings for the direct sound. This area is the product of the normalized sound pressure times the length of the window D. –However here we have divided through by D – so this factor is not shown. The next two integrals represent the total nerve firings for the reverberation – the blue area. –Since we have divided by D, a factor of 1/D is included at the beginning. The second of the two integrals is the physical sum of the sound pressure that would exist if the impulse response was convolved with a steady excitation. The first integral finds the area under this curve. In the second integral we have excluded the direct sound – assuming this will be in the first 5 milliseconds. The limits of the integrals have been adjusted to account for this exclusion. Thus the second integral goes from.005 seconds to the end, and the first integral is from zero to the window width minus.005. I have included the -1.5dB adjustment for my personal thresholds.
Matlab code for LOC % load in a.wav file containing a binaural impulse response – filter it and truncate the beginning upper_scale =20; % 20dB range for firings % proposed box length box_length = round(100*sr/1000); % try 100ms early_time = round(5*sr/1000); D = box_length; %the window width ir_left = data1;% the binaural IR ir_right = data2; clear data1 data2 % filter the Irs wb = [2*1000/sr 2*4000/sr]; [b a] = ellip(3,2,30,wb); ir_left = filter(b,a,ir_left); ir_right = filter(b,a,ir_right); clear data1 data2 wb = [2*1000/sr 2*4000/sr]; [b a] = ellip(3,2,30,wb); ir_left = filter(b,a,ir_left); ir_right = filter(b,a,ir_right); for il = 1:0.1*sr if abs(ir_left(il)) > 500 break end if abs(ir_right(il)) > 500 break end ir_left(1:il) = ; ir_right(1:il) = ; % ir_left is an ipselateral binaural impulse response, %truncated to start at zero and filtered to Hz. % early_time is 5ms in samples, D is 100ms in samples. % here starts the equation on the slide: S = 20-10*log10(sum(ir_left.^2)); early = 10*log10(sum(ir_left(1:early_time).^2)); % first integral is a cumsum representing the build up in %energy when the IR is excited by a steady tone: ln = length(ir_left); log_rvb = 10*log10(cumsum(ir_left(early_time:ln).^2)); % look at positive values of S+log_rvb only for ix = 1:ln-early_time if S+log_rvb(ix) < 0 log_rvb(ix) = -S; end LOC = S-1.5+early -(1/D)*sum(S+log_rvb(1:D-early_time))
Use of the localization equation Just as RT or C80, LOC uses a measured impulse response as an input, with the direct sound starting at time zero. This is the only data a user needs to supply. –The measure is calibrated for a front facing binaural impulse response. An omnidirectional impulse response will give lower values of LOC for the same seat position, due to the lack of head shadowing. The localization equation appears more complex than most current measures for room acoustics, but it has a simple, physiologically based interpretation. –It is the ratio in dB of the number of nerve firings received by the brain from the direct sound in a 100ms window, divided by the number of nerve firings received from all reflections in the same time period. –It contains three experimentally based parameters: the window width D, the dynamic range of the nerve channels S, and the time window for separating direct sound from reflections (5ms). These parameters are not intended to be adjustable without further experimental work. –Matlab code for calculating LOC is simple, as can be seen above.
Interpretation of LOC LOC was developed and verified as a method for predicting when a sound will be accurately localized when the direct sound is much lower in total energy than the sum of all reflections. Like C80, IACC80, and similar measures, LOC is based on a time window that begins with the onset of the direct sound. –In practice, syllables or notes that will be affected by any of these measures will depend on the rise time (onset time) of the sound. –If the sound starts gradually the precise moment of onset becomes indeterminate, and separating direct sound from reflections becomes impossible. –Thus LOC – and other such measures – are accurately predictive only for signals with sharp onsets. –Additionally, if the direct sound from a note or syllable is masked by reverberation from a previous sound, the direct sound will not be audible. LOC predicts the audibility of the direct sound for a syllable or note with a rapid rise-time when there is sufficient freedom from masking from previous sounds. –Although musical signals often do not meet these criteria, in practice there are enough occasions that do meet the criteria that the LOC equation is useful. Remember that for the purposes of this talk Localization is only a proxy for the main goal – predicting when the direct sound is sufficiently audible to produce engagement. –Preliminary results suggest LOC achieves this goal.
Localization Equation Setup The Localization Equation was developed and tested using binaural impulse response generated using the authors own HRTFs. –The source position was 15 degrees to the left (and right) of center. Only the ipselateral channel was analyzed. –Male speech alternated from left to right with a time gap of 400ms, to allow for complete decay of the reverberation between each word. –The reverberation was generated using an independent decaying noise signal convolved with each of 54 HRTFs spaced equally around the listening position. –The HRTFs were equalized so that the azimuth zero elevation zero HRTF was flat from 40Hz to about 4kHz. The elevation notch at 7.8kHz was not equalized away, but was left in place. –Playback was done through headphones equalized to match a loudspeaker placed in front of the listener – again not equalizing the 7.8kHz notch from the listeners frontal HRTF of the loudspeaker. Because my data show that the perception of both localization and near/far is mostly a high frequency phenomenon, the impulse response was bandpass filtered between 700Hz and 4000Hz before being analyzed for localization. –If a measured binaural impulse response is used as an input, care should be taken to insure the dummy head is equalized as described above. –Because of the importance of upward masking in localization, if the low frequencies in the room signal are significantly stronger than those in the frequency range from 700 to 4000Hz, localization is likely to be poorer than the equation would predict.
Comments on LOC –LOC is based on the LOG of the build-up of reverberant energy. This follows directly from the physiological model. Current measures integrate the sound energy rather than the log of sound energy. But our physiology works differently. One of the consequences is that reflections that arrive early have more influence than reflections that arrive later. –As energy builds up additional reflections are not counted as strongly. Reflections later than 100ms are ignored in calculating LOC. –This is very different from C80 or C50, which count the earliest reflections a part of the direct sound, and compare the energy sum to the energy sum of all the later reverberation. In a small hall most of the energy arrives before 80ms regardless of the relative strength of the direct sound, so C80 and C50 are usually high. But small halls can have high C80 or C50, poor localization, and a lack of clarity. –LOC depends strongly on the delay between the direct sound and the build-up of the reverberation. late reverberation does not impair localization of short notes. The principle difference between the localizability in small halls and large halls is the rate at which reflected energy builds up after the start of a note. –LOC is NOT related to EDT – even if Jordans original definition of EDT is used. EDT is relatively independent of the initial time delay When D/R < -10dB, EDT and RT are the same, as there is insufficient direct sound to be detected in a reverse integrated impulse response. –LOC correlates with IACC80 – but IACC is not sensitive to medial reflections. IACC is sensitive to the sum of reflected energy – not the log of energy, and thus is insensitive to when the reflections arrive
Tests with speech A speech signal was convolved with a pair of binaural impulse responses, such that the sound appears to come from +-15 degrees from the front. Then a fully spatially diffuse reverberation was added, in such a way as the D/R, the RT, and the time delay before the reverberation onset could be varied.
Broadband Speech Data Blue – experimental thresholds for the alternating speech with a 1 second reverb time. Red – the threshold predicted by the localization equation. Black – experimental thresholds for RT = 2seconds. Cyan – thresholds predicted by the localization equation.
Threshold Data from Other Subjects – 1s RT Seven subjects participated in a threshold experiment at Kyushu University. –In these experiments the threshold was defined by the extinction of localization, not by the reduction of angle by a factor of two. –Consequently the thresholds are lower than they were in my previous experiment, and they have more variation. However, the data is consistent to within 3dB Blue – new data using absence of any localization as a criterion for threshold. Red – the authors previous data based on a half-angle criterion.
Analysis Add reverb at 2s RT -10dB D/R Two second RT at -10dB D/R does not seriously interfere with pitch.
Add reverb at 1s RT -10dB D/r RT = 1s is much more damaging to harmonic coherence in the 100ms window.
An existing small hall – pictures Note the highly reflective stage and side walls, deeply coffered ceiling, and relatively low internal volume per seat. The sound in many seats is muddy. Adding reflections or decreasing absorption only increases the muddiness.
Hall data The pictures show a recital hall of cubic feet (1840 cubic meters). Designed for 350 seats, it has currently 300 seats, giving a volume/seat of 6 cubic meters. There is 1400 square feet of carpet under the seats on the floor. Reverberation Time (RT) unoccupied is 1.1 seconds from 1000Hz to 63Hz. C80, dominated by the reverberation time, is ~+5dB everywhere. The parallel side walls of the stage provide little diffusion. The hall is generally liked by the audience and players, although there are reports of loudness and balance problems on stage. Musicians desire more resonance and greater clarity in the middle of the hall.
Experiments with absorption and acoustic enhancement Measurements and experiments involving various combinations of fiberglass panels and electronic reverberation enhancement were conducted in January –Measurements were made with three loudspeakers, three dummy heads, and a Soundfield microphone. –All musical performances were recorded with the same microphones, and with an array of close microphones on stage. About 30 musicians participated, including faculty, staff, students from all three divisions, and musicians from the wider community. The goal was to improve the instrumental balance on stage, reduce excess stage loudness, and to increase resonance and the ability to hear individual instruments throughout the hall. With both panels and enhancement in place comments from the participants were favorable. Players and singers found balancing with piano was easier, and the middle registers of the piano were more easily heard both by the musicians and in the hall.
The absorptive curtains at the rear of the stage could be rapidly withdrawn. The blankets that simulated audience could be removed in 5 minutes, along with the panels on stage. This allowed prompt A/B comparisons. Some of the 25 LARES enhancement speakers are visible
Results from the experiments The experiments in January showed that adding fiberglass panels around the stage increased clarity and the ability to localize instruments in the hall, raising the measured value of LOC from an average of minus 1.5dB to +3dB or more. Localization and clarity in the balcony were additionally improved by adding panels to the upper audience right side wall, which eliminated the strong lateral reflection from that surface. –The lower surface of this wall was already absorptive. The electronic enhancement successfully compensated for the loss of resonance due to the panels. Without the enhancement the perceived resonance was reduced. In a subsequent experiment with a violin-piano combination and no enhancement we found that just 12 fiberglass panels each 2x6x2 around the bottom of the stage noticeably improved the clarity on the floor of the hall, and also improved the balance for the players on stage. For this music the reduced resonance was not a problem. –These panels absorbed the first-order reflection from the back of the stage, which has the highest level and the shortest time delay. Absorbing this reflection contributed strongly to increasing LOC.
Usefulness of the measure LOC LOC informs us that the primary contribution to difficulty in localization are the first strong reflections, regardless of the direction they come from. We initially thought that since the floor of the hall is not a significant source of these reflections, it is would be likely that removing the carpet under the seats would raise the RT without decreasing LOC significantly. –However LOC is also sensitive to reverberation which arrives before 100ms, and this would be increased by removing the carpet. –A few later experiments suggest that removing the carpet will increase the reverberant level sufficiently to eliminate the improvement in LOC provided by the absorption on stage. The existence of a LOC as a physical measure can help to answer these questions in advance – or at least suggest that an experiment is needed before drastic alterations are undertaken.
Small shoebox halls can be OK If the client insists on a shoebox it can work by building a large hall and installing a small number of seats. –I was just in such a small hall in Helsinki, and at least half the seats were OK. But this is not the ideal solution. –With a different shape nearly all the seats could have been OK – and it might have been less expensive.
Light models I ran across these pictures while cleaning out my office. The top one is a too-simple model of the Philadelphia Academy of Music. The bottom is intended to be BSH, but with a single balcony. I abandoned light modeling because it does NOT provide any information about the time delay gap – nor information about the effects of note length on D/R. But it DOES provide information about the total reverberant energy compared to the direct. And very complex hall shapes can be quickly modeled.
A few slides from earlier measures of azimuth from recordings, using running IACC in a 10ms window. They are followed by an earlier measure of harmonic coherence.
Localization The figure shows the number of times per second that a solo violin can be localized from row 4 of a small shoebox hall (~500 seats) near Helsinki. It also shows the perceived azimuth of the violin As can be seen, the localization – achieved at the onsets of notes – is quite good, and the azimuth, ~10 degrees to the left of center, is accurate.
Localization – surface1 Here we plot the same data for the violin as a function of (inverse) azimuth, and the third octave frequency band. As can be seen, for this instrument the principle localization components come at about 1300Hz. Interestingly, Human ability to detect azimuth, as shown in the threshold data, may be maximum at this frequency.
Localization, Surface 2 Here we plot 1/(1-IACC) as a function of time and third octave band. Note that the IACC peaks at the onset of notes can have quite high values for a brief time. This happens when there is sufficient delay between the direct and the reverberation, and sufficient D/R.
Localization – a poor seat Here is a similar diagram for a solo violin in row 11 of the same hall. The sound here is unclear, and the localization of the violin is poor. As can be seen, the number of localizations per second is low (in this case the value really depends on the setting of the threshold in the software). Perhaps more tellingly, the azimuth detected seems random. This is really just noise, and is perceived as such.
Measures based on harmonic coherence In the absence of reflections the formant frequencies above 1000Hz are amplitude modulated by the phase coherence of the upper harmonics. This modulation is easily heard, creating the perception of roughness (Zwicker). –Reflections randomize the phase of these harmonics. The result is highly audible, and is a primary cue for the distance of an actor, singer, or soloist. This effect can be measured with live recordings, and is sensitive both to medial and lateral reflections. This graph shows the frequency and amplitude of the amplitude modulation of a voice fundamental in the 2kHz 1/3 octave band. The vertical axis shows the effective D/R ratio at the beginning of two notes from an opera singer in Oslo to the front of the third balcony (fully occupied.) The sound there is often muddy, but the fundamental pitch of this singer came through strongly at the beginning of these two notes. He seemed to be speaking directly to me, and I liked it.
Another singer From the same seat the king (in Verdis Don Carlos) was not able to reach the third balcony with the same strength. Like the localization graph shown in a previous slide, this graph seems to be mostly noise. The fundamental pitches are not well defined. The singer seemed muddy and far away. His aria can be heart-rending – but here it was somewhat muted by the acoustics. We were watching the king feel powerless and forlorn. But we were not engaged.