Abstract:
A method of increasing the intelligibility of an audio broadcast in an at least partially enclosed space from at least one amplified audio source. An input microphone receives an incident audio wavefront at a first position in the at least partially enclosed space. An active noise control system is employed to generate a cancelling audio wavefront having a magnitude substantially equal to the magnitude of incident audio wavefront and a phase substantially opposite to the phase of the incident audio wavefront. The cancelling audio wavefront is broadcast at a second position in the at least partially enclosed space adjacent to a reflective surface of the at least partially enclosed space so as to attenuate the incident audio wavefront substantially at or near the reflective surface in order to reduce reverberations of the incident audio wavefront. In this manner, reverberations which could reduce the intelligibility of the audio broadcast to an audience is reduced.

Description:
FIELD OF THE INVENTION 
     The invention relates generally to the art of active noise cancellation, and more particularly to a method and system of reducing acoustical reverberations in an at least partially enclosed space. 
     BACKGROUND OF THE INVENTION 
     In a public venue such as a large auditorium, excessive reverberation negatively impacts the sound quality of audio material being delivered to an audience. Reverberation is the persistence of sound in a particular space after the original sound dissipates. A reverberation, or reverb, is created when a sound is produced in an enclosed space causing a large number of echoes to build up and then slowly decay as the sound is absorbed by the walls and air. If the ratio of source audio to reverberation is low enough, the source material is masked by the presence of the reflected/reverberant audio. This leads to degradation in the quality of the presented source material, leading to difficulty in understanding speech. If the reverberation time is too long, the presented material sounds ‘muddy’. These problems are exacerbated in larger auditoriums, since the larger the auditorium, the longer the delay and the greater the reverberation time. 
     Ideally, an auditorium should be designed for good acoustics. The shape of the auditorium is a major factor in the ratio of direct to reflected sound arriving at the audience. Sound absorbing material on the walls and ceilings can reduce the amount of sound reflected around an auditorium. Diffuse reflectors placed around the ceilings and walls help prevent the reflections from being too coherent, so that an ‘echo’ is not perceived. 
     However, relatively few public venues are actually designed for ‘good’ acoustics. Music concerts and other public events are usually held in the largest venue possible such as sports arenas because the performer typically wishes to sell as many tickets, and fill as many seats, as possible. However, in these large venues there is typically a severe degradation of the perceived audio due to the delayed reflections and reverberations of the source audio from the walls and ceilings. 
     One of the simplest techniques of reducing the effects of auditorium reverberation is to elevate the loudspeakers and point the sound in the direction of the audience, and away from the ceiling of the auditorium. This reduces the amount of sound pointed directly at a reflective ceiling and back wall, reducing reflection and hence, reverberation, from these surfaces. 
     However, this is not always possible, especially in the case of a large venue. Further, if the sound source is perceptively disjointed with the source (singer, speaker, etc . . . ), this can have a negative impact on the audience. 
     Advancements in electronics have lead to the development of phased array loudspeakers, wherein the timing of sounds arriving at the various loudspeakers in the array allow the source audio to be ‘steered’ towards the audience. 
     Another approach is to place loudspeakers throughout the auditorium, pointed at the audience. The human acoustic perception of the source of sound is maintained by delaying the sound at remote loudspeakers so that it arrives at the audience within milliseconds after the sound from the stage. 
     However, these solutions are not always practical, given the shape and size of a particular venue. 
     Another approach is based on active sound cancellation, in which a particular sound or noise is monitored and a phase-shifted sound is generated to cancel or reduce the unwanted sound or noise. 
     For example, U.S. Pat. No. 5,363,451 describes a system and method for cancelling noise in an open space from a particular noise source such as a ‘machine’. The system detects noise at the noise source and cancels it at another point. 
     U.S. Pat. No. 5,699,437 describes a system that employs arrays of microphones to pick up ‘noise’ emanating from some point inside an auditorium and plays the ‘opposite sound’ or ‘cancelling noise’ from loudspeakers situated around the auditorium such that the noise is cancelled at some optimal location in the auditorium (the audience). Note that this solution can improve the sound at a ‘sweet spot’ in the auditorium, but it can also result in detrimental performance in another area of the same auditorium. Further, this system needs tuning by an expert on installation. 
     U.S. Pat. No. 6,795,557 describes a method for reducing the level of acoustical reflections in a room by attempting to cancel reflected signals as they go into the intended ‘quiet area’. However, at this point it becomes difficult to distinguish between reverberations and source audio, particularly where the room or venue has multiple reflective surfaces where the reverberations are not coherent. 
     SUMMARY OF THE INVENTION 
     Generally speaking, the invention attenuates sound at or near a reflection surface in the room in order to prevent or at least reduce the sound from reflecting back into a listening area. The invention thus differs from the prior art where no attempt is made to stop reflections and a ‘quiet zone’ is created by attenuating sounds that do reflect into the listening area. 
     According to one aspect of the invention a method is provided for increasing the intelligibility of an audio broadcast in a room from at least one amplified audio source. The method includes: receiving an incident audio wavefront from the amplified audio source at a first position in the room; generating a cancelling audio wavefront having a magnitude substantially equal to the magnitude of incident audio wavefront and a phase substantially opposite to the phase of the incident audio wavefront; and broadcasting the cancelling audio wavefront at a second position in the room, the second position being adjacent to a reflective surface of the room and downstream of the first position relative to the audio source. Thus, the incident audio wavefront is attenuated substantially at or near the reflective surface in order to reduce reflections of the incident audio wavefront. 
     The incident audio wavefront can be received via at least one input microphone. The cancelling audio wavefront can be broadcast through at least one directional loudspeaker positioned adjacent to, and facing, the reflective surface. 
     The cancelling loudspeaker is preferably spaced at a distance D away from the input microphone, where D corresponds to the delay time required to generate the cancelling audio wavefront. 
     To achieve broad coverage, an array of input microphones and corresponding cancelling loudspeakers can be deployed across the breadth of the reflective surface. The array can involve a plurality of independent single-channel sound dampening units, or the components can be interlinked via a multiple-channel active sound control module. 
     According to another aspect of the invention another method of increasing the intelligibility of an audio broadcast in a room from at least one amplified audio source includes the steps of: deploying at least one input microphone in the room to convert at least one incident audio wavefront into at least one corresponding reference electrical signal; deploying at least one cancelling loudspeaker adjacent a reflective surface in the room to broadcast at least one secondary audio wavefront; deploying at least one error microphone substantially at the reflective surface to convert a superposition of the at least one incident audio wavefront and the at least one secondary audio wavefront into at least one error electrical signal; generating at least one cancellation signal based on the at least one reference electrical signal so as to minimize the power of the at least one error electrical signal; and converting the cancellation signal into said at least one second audio wavefront. 
     In the application of this method, each cancelling loudspeaker can be positioned adjacent to the reflective surface and may have a highly directional output directed downstream of the incident audio source. Each cancelling loudspeaker is preferably also spaced at least a distance D away from a corresponding input microphone, where D corresponds to the delay time required to generate the cancelling audio wavefront after the incident audio wavefront is picked up by the input microphone. 
     For broad coverage, an array of input microphones, an array of cancelling loudspeakers, and an array of error microphones can be deployed across the breadth of one or more reflective surfaces in the room. 
     In one embodiment, the number of input microphones, cancelling loudspeakers and error microphones can be equal, and each input microphone, cancelling speaker and error microphone triplet form an independent, single-channel, sound dampening unit, with a plurality of single-channel sound dampening units being deployed across the breadth of the reflective surface(s). 
     In another embodiment the input microphones, cancelling loudspeakers and error microphones can be interlinked via a multiple-channel active sound control module. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
       The foregoing and other aspects of the invention will be more readily appreciated having reference to the drawings, wherein: 
         FIG. 1  is a schematic diagram of a reverberation control system according to a first embodiment; 
         FIG. 2  is an electronic system block diagram of a control module employed in the system shown in  FIG. 1 ; 
         FIG. 3  is a logical block diagram of a controller employed by the system shown in  FIG. 1 ; 
         FIG. 4  is a schematic diagram of a reverberation control system according to a second embodiment; 
         FIG. 5  is a schematic diagram of a reverberation control system according to a third embodiment; 
         FIG. 6  is a logical block diagram of a controller employed by the system shown in  FIG. 5 ; and 
         FIG. 7  is a schematic diagram of a reverberation control system according to a fourth embodiment. 
     
    
    
     DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS 
       FIG. 1  shows a reverberation control system  100  comprising one sound dampening unit  102 . The unit  102  includes an input microphone  110 , a noise cancellation loudspeaker  112 , and an error microphone  114 , all connected to a digital control module  116 . 
     Importantly, the sound dampening unit  102  is installed between an audio source  104  and an acoustic reflective surface  106  such as the rear wall or ceiling of an at least partially enclosed space such as a room, an auditorium, a concert hall, a religious venue (eg. a church, a synagogue, a mosque), a sports arena and a stadium. More particularly, the unit  102  is installed near the acoustic reflective surface  106  to dampen sound travelling toward the reflective surface  106  before the sound reflects from the reflective surface  106  to become an echo or contribute to reverberation. Thus, the invention attempts to remove reflections at, or very near, the point of reflection without deleteriously affecting the sound in the bulk of the space. 
     Accordingly, input microphone  110  detects an incident audio wavefront  108  travelling towards the reflective surface  106  and feeds an electronic signal representative of the incident audio wavefront  108  to the control module  116 . The control module calculates a cancelling audio wave  118  that is substantially of the same amplitude but opposite in polarity (i.e., equal amplitude, opposite phase) to the incident audio wavefront  108 . The control module  116  plays the cancelling audio wave  118  through the loudspeaker  112 . The loudspeaker  112  is preferably a highly directional audio source as known in the art per se that faces the reflective surface  106  so as to highly attenuate the incident audio wavefront  108  at or near the reflective surface  106  without excessive bleeding of the cancelling audio wave  118  back towards the audio source  104  or the audience. The error microphone  114  is used to sample the sound substantially at the reflective surface  106  and provide a feedback signal to the control module  116 . 
     The control module  116  executes a noise cancellation algorithm that preferably outputs the cancelling audio wave signal within a set delay time T (between the time the input microphone senses the incident wave and the loudspeaker plays the cancellation wave). With this knowledge, the input microphone  110  is preferably spaced a distance D away from the loudspeaker  112  corresponding to time T required for the incident audio wavefront  108  to travel between these components. Depending on the noise cancellation algorithm employed, the delay time T may also be adaptively tuned based on the feedback provided by the error microphone  114 . (The delay time may be adjusted in the algorithm that adaptively computes the weights of digital filter taps, as such algorithms are discussed in greater detail below.) 
     The sound dampening unit  102  may be assembled onto a rigid frame bolted onto the reflective surface  106 . Alternatively, the components of the unit may be independently mounted in the at least partially enclosed space. For example, the components may be suspended from the ceiling. 
       FIG. 2  is a system block diagram of an exemplary version of the control module  116 . The input microphone  110  is connected to a preamplifier  122  which optionally feeds into a first gain stage  124  for maintaining overall system dynamic range. The error microphone  114  is likewise connected to a preamplifier  126  which feeds into a second gain stage  128 . The input and error microphone signals are then applied to an audio coder/decoder (‘codec’)  130  for converting the analog microphone signals into digital input and error audio signals. The audio codec  130  is connected via a high speed connection to a digital signal processor (DSP)  132 . An example of suitable integrated circuits for the DSP and codec is the TMS320C6472 DSP and TLV320AIC3254 codec from Texas Instruments. A high speed random access memory  134  for computational purposes and a flash memory  136  for storing boot programming and other purposes may be connected to the DSP  132 . The DSP  132  executes the sound cancellation algorithm and provides the cancelling audio wave signal to the audio codec  130  which feeds a power amplifier  140  that drives the loudspeaker  112 . 
     From a control point-of-view, the sound dampening unit  102  may be implemented as a single channel broadband adaptive feedforward control system having a single reference sensor, a single secondary source and a single error sensor. A suitable control system which takes into account secondary path effects S(z) resulting from the electronic circuitry and the acoustic path from loudspeaker to the error microphone where the incident wavefront is combined with the output of loudspeaker is shown in  FIG. 3 . In this model, x(n) is the incident audio wavefront  108  as sampled by the input microphone  110 . The summing junction represents the superposition of the cancelling audio wave  118  ( FIG. 1 ) from loudspeaker  112  with the incident audio wavefront  108 , with e(n) ( FIG. 3 ) representing the error in the superpositioned sound. W(z) is an adaptive filter implemented by the DSP  132  that is used to estimate an unknown space P(z) representing the primary acoustic path from the input microphone  110  to the error microphone  114 , and y(n) is the output of the adaptive filter. S(z) is the secondary path transfer function. The aim of the adaptive filter is to minimize the power of the error signal e(n), and a least mean square (LMS) block implemented by the DSP  132  adaptively modifies the weights of W(z). 
     If desired, the feedback caused by the cancelling audio wave  118  being picked up the input microphone  110  may also be neutralized by the use of a separate feedback cancellation filter. See generally, Kuo and Morgan, “Active Noise Control: A Tutorial Review”, Proceedings of the IEEE, Vol. 87, No. 6, June 1999, the contents of which are incorporated herein by reference in their entirety. 
     An alternative noise cancellation algorithm that may provide a more stable system includes the filtered-X LMS algorithm (Widrow and Stearns, 1985). Faster converging algorithms include the recursive least square algorithm (Kuo and Morgan, 1996), and neural network based techniques. 
       FIG. 4  shows a variant  200  of the reverberation control system shown in  FIG. 1 . In the reverberation control system  200 , multiple sound dampening units  102  are deployed along and proximate to the reflective surface  106 , such as a wall or ceiling. Each unit  102 , however, is a single channel system, functioning essentially independently of every other unit  102 . 
       FIG. 5  shows another more preferred embodiment of a reverberation control system  300  that employs a multiple-channel active sound control module that interlinks several input microphones  310 , several cancelling loudspeakers  312  and several error microphones  314  proximate a reflective surface  306 . This system can be deployed in larger auditoriums, venues or halls where there may be several audio sources  304  broadcasting sound, resulting in a more complex incident wavefront  308 . 
     The illustrated multi-channel reverberation control system  300  employs J input microphones  310  to form a reference signal vector. The system generates K cancelling signals to drive the cancelling loudspeakers  312 , and utilizes M error microphones  314  distributed along the area of the reflective surface(s) to measure the residual sound. 
     A logical block diagram of a suitable controller  320  is shown in  FIG. 6 . The wide arrows represent an array of electrical or acoustic signals that are symbolically expressed as vectors. The matrix P represents M×J primary path transfer functions, P mj (z), from the audio source to each error microphone output, E m (n). The matrix S represents M×K secondary-path transfer functions, S mk (z), from the K cancelling loudspeakers  312  to the M error microphones  314 . The controller  320  also incorporates feedback paths from the cancelling loudspeakers  312  to the input microphones and matrix F represents J×K feedback paths, F jk (z), from the K cancelling loudspeakers  312  to the J input microphones  310 . There are thus K×J possible feedforward channels, each having a separate adaptive filter, the K×J adaptive filters W k,j (z) being represented by the matrix W. A multiple-reference/multiple-output filtered-X LMS algorithm is employed for adaptively determining the weights of the filter taps. 
     Alternative controllers include those that employ subband techniques where the adaptive weights are computed in subbands but are then collectively transformed into an equivalent set of wideband filter coefficients. Processing the signals in subbands can reduce the computational load and result in faster convergence of filter weights because the frequency range is reduced in each subband. See Morgan and Thi, “A delayless subband adaptive filter”, IEEE Trans. Signal processing, vol. ASSP-32, pp. 304-337, April 1984. 
     As will be seen, the reverberation control systems  100 ,  200 ,  300  are deployed to attenuate sound at or near a reflection surface in order to prevent or at least reduce the sound from reverberating back into a listening area. The invention thus differs from the prior art that attempted to create a ‘quiet zone’ by attenuating directly incident sounds plus reverberations from the sounds that do reflect into the listening area. In the ‘quiet zone’ application, the digital filter may require a large number of taps in order to deal with the reverberations. This increases the computational load on the DSP, necessitating either a more powerful DSP processor or limiting the type of noise cancellation algorithms to those having relatively low computational requirements, such as delayless subband adaptive filters. However, the applications discussed herein that attempt to stop or limit reverberations before they become a problem for the listener will likely result in a digital filter requiring a comparatively smaller number of taps. 
     From the foregoing it will be appreciated that the reverberation control system can be applied in a variety of environments. A concert hall is one environment. Churches with high ceilings that tend to have many reflective surfaces are another environment. Other environments include bus and train stations where reverberations can make announcements unintelligible. The invention can also be applied in factory environments to reduce machine noise reflected from the ceilings or walls around a noisy machine to reduce the overall noise in the factory. 
     Another advantage of providing the reverberation control system relates to the environment outside the venue in which the system is installed. Without a reverberation control system such as that which has been described herein, some sound energy is transmitted through the structure of the venue itself and out into the surrounding environment (eg. the surrounding neighbourhood), which may be undesirable to people in that surrounding environment. Because the reverberation control system reduces the amount of sound energy that reaches the structure of the venue, the amount of sound that is transmitted out of the venue is reduced, and as a result, surrounding buildings, houses and the like are not subjected to as much noise. Furthermore, the structure of the venue itself may incur a reduced amount of sound wave-initiated damage (eg. cracks in an interior façade due to fatigue resulting from the impact of sound waves thereon) as a result of the reduced amount of sound energy that reaches the structure. 
     In another embodiment shown at  500  in  FIG. 7 , the same reference numerals as shown in  FIG. 1  are used to indicate analogous components. A primary difference between the embodiment in  FIG. 7  and the embodiment shown in  FIG. 1  is that in the embodiment in  FIG. 7 , the control module  116  receives audio signals via conduit  502  from audio signal source  504 , which is the audio signal source that feeds the audio source  104  through conduit  506 , instead of receiving signals from an input microphone. The control module  116  processes the audio signal from conduit  502  to generate a cancelling audio wave  118  via loudspeaker  112  that at least in part cancels the incident audio wave  108 . The incident audio wave  108  is made up largely of the source audio (ie. the audio that is emitted from the speaker  104 ), but may also include other audio such as, for example, audience noise, ventilation noise and audio reflected from other surfaces of the venue. The system  500  uses the error microphone  114  similarly to the embodiment shown in  FIG. 1  (ie. to sample the sound substantially at the reflective surface  106  and provide a feedback signal to the control module  116 ). 
     While the above describes particular embodiments of the invention, it will be appreciated that modifications and variations may be made to the detailed embodiment described herein without departing from the spirit of the invention.