Abstract:
An acoustic tracking system that uses an array of microphones to determine the location of an acoustical source. Several points in space are determined to be potential locations of an acoustical source. Beamforming parameters for the array of microphones are determined at each potential location. The beamforming parameters are applied to the sound received by the microphone array for all of the potential locations and data is gathered for each beam. The most likely location is then determined by comparing the data from each beam. Once the location is determined, a camera is directed toward this source.

Description:
BACKGROUND OF THE INVENTION  
         [0001]    1. Field of the Invention  
           [0002]    The invention relates generally to videoconferencing systems and more specifically to camera tracking of videoconference participants.  
           [0003]    2. Description of Related Art  
           [0004]    When people wish to communicate from remote locations, a videoconferencing system is a convenient way to share ideas. Typically, there is more than one person participating in a videoconference. A camera must, therefore, be positioned to frame everyone in the room. However, such camera shots are impersonal and do not allow the recipient of the image to pick up on nuances of the speaker&#39;s facial expressions because they lack sufficient detail. Manually tracking the camera from one speaker to the next is inconvenient and distracting. There are a number of prior art patents in this field directed to solving this problem.  
           [0005]    For example, U.S. Pat. No. 5,686,957 to Baker teaches automatic tracking by using a plurality of microphones and a crude audio detection circuit. Because the audio detection mechanism is limited in its ability to locate sound, a special camera is used to enhance the peripheral portion of the field of view. The camera rotates in approximately 30 degree increments in order to frame the general location of the speaker.  
           [0006]    U.S. Pat. No. 5,778,082 to Chu teaches automatic tracking by using a plurality of microphones and a processor. The processor determines when an acoustic signal begins at each microphone, and then locates the direction of the source based on a comparison of data from each microphone. The system, however, requires processor intensive Fast Fourier Transform calculations to be made continuously.  
           [0007]    U.S. Pat. No. 5,940,188 to Van Schyndel teaches automatic tracking using an optical transducer that takes visual cues (e.g., a moving mouth) to point the camera toward the location of the speaker. The method requires a very advanced processor, an optical transducer, and is subject to many false signals (e.g., one participant whispering to a neighbor).  
           [0008]    U.S. Pat. No. 5,959,667 to Maeng teaches automatic tracking using a microphone array and a set of preset values. The preset values include a set of camera parameters that define a particular camera shot. The microphone array locates the position of the speaker, and the position is compared to the preset values. The camera then tracks to the closest preset position. The method also requires a very powerful processor because the speaker location has to be continuously calculated.  
           [0009]    U.S. Pat. No. 6,005,610 to Pingali teaches a method for localizing a speaker based on concurrent analysis of a video image and plural microphone signals. The position of the speaker is tracked as the speaker changes position.  
           [0010]    U.S. Pat. No. 5,438,357 to McNelley teaches a method for image manipulation that creates the impression of eye contact between video conferencing parties. The camera is manipulated to track a speaker such that their image is centered on a display.  
           [0011]    U.S. Pat. No. 5,528,289 to Cortjens et al. teaches a system for controlling devices on a computer network. The pan, tilt, and zoom of each camera is controlled via a pointing device such as joystick or mouse.  
           [0012]    U.S. Pat. No. 5,581,620 to Brandstein et al. teaches a method for enhancing the reception of signals received at several locations using beamforming techniques. The location of a speaker is calculated and the information is used to align the phase of signals from multiple audio detectors. The signals originating from the speaker&#39;s location are added while signals from other locations are attenuated.  
           [0013]    U.S. Pat. No. 5,583,565 to Cortjens et al. teaches a camera system that can be remotely controlled over a computer network for the purpose of video conferencing. Preset operational parameters are stored within the camera to increase the ease of operation. A user input device is used to control the camera.  
           [0014]    U.S. Pat. No. 5,844,599 to Hildin teaches a video system in which a camera is directed at a voice activated emitter associated with a speaker. The emitters are detected using an infrared position signal and a video camera is directed at the emitter&#39;s location.  
         BRIEF SUMMARY OF THE INVENTION  
         [0015]    In one embodiment, the invention provides an acoustic tracking system for determining the location of a sound producing acoustical source from a set of at least two user-defined locations. The system includes an array of microphones and a processor for performing various calculations. The processor can be any device that can make the necessary calculations such as a CPU, FPGA, and the like.  
           [0016]    The calculations result in a set of beamforming parameters, a set of modified signals, and a result location. Each beamforming parameter within the set of beamforming parameters is associated with each user-defined location within the set of at least two user-defined locations. The set of modified signals is determined by applying each beamforming parameter within the set of beamforming parameters to the sound received by the array of microphones. The result location is then associated with the best modified signal within the set of modified signals. The best modified signal can be determined as a function of signal strength and duration.  
           [0017]    In another embodiment, the invention provides a method for identifying the location of an audio source. The method includes the steps of defining at least two points where the audio source may be located, calculating beamforming parameters for each of the at least two points in space, receiving sound by a microphone array, applying each set of the beamforming parameters to the sound received by the microphones in the array, determining which set of beamforming parameters maximize the sum amplitude of the signals received by the microphone array; and identifying one of the at least two points in space associated with the set of beamforming parameters that maximize the sum amplitude of the microphone signals.  
       
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0018]    [0018]FIG. 1 is a top view of a video conference room including an embodiment of the present invention;  
         [0019]    [0019]FIG. 2 is a mathematical representation of two audio sources generating a signal;  
         [0020]    [0020]FIG. 3 is a mathematical representation of three microphones receiving an audio signal generated from one audio source;  
         [0021]    [0021]FIG. 4 is a flowchart of a preferred method of operation of the invention;  
         [0022]    [0022]FIG. 5 is a diagram of a videoconferencing system and two pre determined points in space;  
         [0023]    [0023]FIG. 6A is a mathematical representation of three microphones receiving a signal in the time domain;  
         [0024]    [0024]FIG. 6B is a mathematical representation of the signal received by the three microphones after they have been adjusted by a first set of beamforming parameters; and  
         [0025]    [0025]FIG. 6C is a mathematical representation of the signal received by the three microphones after they have been adjusted by a second set of beamforming parameters;  
     
    
     DETAILED DESCRIPTION OF THE INVENTION  
       [0026]    [0026]FIG. 1 shows a conference room  100 . Room  100  has a table  110 , three chairs  120 ,  124 , and  128 , a wall  130 , and a videoconferencing system  140 . Videoconferencing system  140  includes a microphone array  150 , a video camera  160  and a movable microphone  170 .  
         [0027]    Movable microphone  170  receives the sound that is transmitted as the audio component of the teleconference to a remote location. Movable microphone  170  is typically kept away from video camera  160  so it does not pick up extraneous noise from the normal operation of video camera  160 .  
         [0028]    In one embodiment of the invention, microphone array  150  is positioned proximate to video camera  160 . As will be described, microphone array  150  consists of at least two microphones, although a typical system would includes five or six microphones. Additionally, if the location of movable microphone  170  relative to videoconferencing system  140  was known, microphone  170  could also be configured to act as part of microphone array  150 . In order to simplify the description of the invention all references to a microphone or microphones in this specification will refer to the microphones within microphone array  150 .  
         [0029]    Three potential audio sources  180 ,  184 , and  188  are shown in room  100 . Sound waves  190  are formed when one of potential audio sources  180  generates sound (i.e., a person who is sitting in chair  120  speaks). As will be described in connection with FIG. 2, videoconferencing system  140  can determine the location of an audio source by analyzing information gathered by microphone array  150  through the use of beamforming techniques. However, as sound waves  190  propagate from audio source  180 , interference such as reverberation  195  off wall  130  may occur, causing error in beamforming analysis. Techniques are known in the art to reduce or eliminate such unwanted reverberation  195  in both the time domain and the frequency domains. Such techniques may be easily included in the systems and methods discussed herein without altering the scope and spirit of the invention.  
         [0030]    [0030]FIG. 2 illustrates an example of beamforming. An audio source  210  generates a signal that propagates radially in three dimensions. The signal travels according to the relationship r=S t, where S is the speed of sound, t is the time that elapsed since the signal was generated and r is the radius of the sphere that represents all points where the sound can be heard at a time t.  
         [0031]    A first microphone  220 , located an r, distance  230  from audio source  210 , receives the signal at a time t 1 , and a second microphone  240 , located an r 2  distance  250  from audio source  210 , receives the signal at a time t 2 . As will be seen, it is advantageous to describe t 2  in terms of Δt 2 , the difference in time between t 1  and t 2 . Although Δt 2  is a positive number in FIG. 2, Δt 2  can be either positive or negative depending on the relative positions of audio source  210 , first microphone  220  and second microphone  240 . The beamforming calculation also requires a distance D 1    260  between first microphone  220  and second microphone  240 . The result of the beamforming calculation is a beam-line  270 , along which the audio signal must have originated in order to produce the measured Δt 2 .  
         [0032]    By arbitrarily assigning the origin (0,0) of a Cartesian coordinate system to second microphone  240  and orienting the coordinates so first microphone  220  is on the X-axis, beam-line  270  can be described as a function of Δt 2  using the following equations:  
           x ( t   1 )=( D   1   2 +2 S   2   t   1   Δt   2   +S   2   Δt   2   2 )/2 D   1   (1) and  
           y ( t   1   , x )=±[ S   2   t   1   2 −( x−D   1 ) 2 ] 1/2   (2).  
         [0033]    Equations (1) and (2) are derived by applying Pythagoras&#39; Theorem to the two right triangles with the hypotenuses r 1  distance  230  and r 2  distance  250 , noting that r 1 =S t 1  and r 2 =S t 2 , and substituting (t 1 +Δt 2 ) for t 2 . Once Δt 2  is known, the x position and y position will change as a function of t 1 . While FIG. 2 shows beam-line  270  for positive values of y, it should be understood that there is a similar beam-line for negative values of y.  
         [0034]    If the time t 1  that the signal takes to reach first microphone  220  is known, then equations (1) and (2) can be used to reduce the possible positions of audio source to one of two points ((x, y) and (x, −y)). However, since the t 1  time is usually not known, the most information that can be derived from microphone array  150  with only two microphones is that audio source  210  originated along beam line  270 . A third microphone is required to determine the location of audio source  210  along beam-line  270 .  
         [0035]    [0035]FIG. 3 illustrates an example of beamforming using a third microphone  310 . Third microphone  310  is located an r 3  distance  320  from audio source  210 , receives the signal at a time t 3 , and is a D 2  distance  330  from second microphone  240 . By using equations (1) and (2), applying Pythagoras&#39; Theorem to the right triangle with r 3  distance  320  as a hypotenuse, noting that r 3 =S t 3  and substituting (t 1 +Δt 3 ) for t 3 , the equation:  
         [0036]    [0036] t   1 =( D   1   D   2   2   +D   1   S   2   Δt   3   2   −D   2   S   2   Δt   2   2   −D   1   2   D   2 )/(2 D   2   S   2   Δt   2 −2 D   1   S   2   Δt   3 )  (3)  
         [0037]    can be derived.  
         [0038]    By combining the three equations, audio source  210  can be located by knowing D 1  distance  260  (the spacing between first microphone  240  and second microphone  220 ), D 2  distance  330  (the spacing between second microphone  220  and third microphone  310 ), and the Δt 2  and Δt 3  time intervals (the length of time between when second microphone  240  and third microphones  310  receive the signal relative to the time first microphone  220  receives the signal).  
         [0039]    The three microphones  220 ,  240 , and  310  are shown to be co-linear for simplicity. However, it should be appreciated that a non-co-linear arrangement of the three microphones enables identification of a pair of source positions, (x, y, z) and (x, y, −z), in three-dimensional space. The three non-co-linear microphones form a plane, and all points on one side of the plane are represented as positive z and points on the other side are represented as negative z. Although a fourth non-planar microphone can then be used to determine which side of the plane the source is located, all the potential audio sources are normally disposed on only one side of microphone array  150 , so a fourth microphone is not necessary to locate a single point in space. However, additional microphones are desirable to provide error checking and accuracy.  
         [0040]    Using these principals, the flowchart in FIG. 4 illustrates the steps of a preferred embodiment of the invention. First, in step  410 , the potential sources of sound are identified. Although only three potential sources are shown in FIG. 1, the invention can be used with any number of potential audio sources. There are many possible ways to identify potential audio sources. For example, the pan, tilt and zoom coordinates of camera  160  can be obtained by bringing potential audio sources into focus. Alternatively, a signaling device can be positioned next to each potential audio source, or a participant can make a sound during an initiation sequence. A pre-programmed matrix of different points in space can represent potential audio sources. These methods of identifying potential sources are merely illustrative and should not be considered a comprehensive list of methods known in the art.  
         [0041]    After the potential sources are identified, next step  420  in FIG. 4 is to calculate beamforming parameters for each point in space. These parameters only need to be calculated once during initialization. Those skilled in the art will appreciate that, in the time domain, the beamforming parameters can represent the time difference between any pair of microphones, and are not restricted to At values (i.e., the time difference between the first microphone and each other microphone in the microphone array). A “set” of beamforming parameters would represent all the beamforming parameters for a certain point in space. Therefore, if the microphone array had only two microphones, the set of beamforming parameters would consist of a single value. If the microphone array had six microphones, the set of beamforming parameters would include a minimum of three beamforming parameters for each potential audio source (although more beamforming parameters could be used for redundancy).  
         [0042]    As an example of steps  410  and  420 , FIG. 5 shows videoconferencing system  140 , two potential audio sources  510  and  515 , and five microphones  520 ,  522 ,  524 ,  546 , and  528 . Once pan  530 , tilt  540 , and zoom  550  of camera  160  are known for each potential source  510  and  515 , the distance of each potential source  510  and  515  to each microphone  520 ,  522 ,  524 ,  546 , and  528  is determined by applying trigonometric principles. Of course, the coordinates of each microphone  520 ,  522 ,  524 ,  526 , and  528  in relation to camera  160  would need to be either known or calculated.  
         [0043]    If microphone  520  was at location (X, Y, Z) in Cartesian coordinates, one of potential sources  180  was located at (ρ, φ, θ) in spherical coordinates, and the origin (0,0) of both the Cartesian coordinates and the spherical coordinates was the camera, the distance could be found with the following equation:  
         Distance=[( X−ρ  cos θ sin φ) 2 +( Y−ρ  sin θ sin φ) 2 +( Z−ρ  cos φ) 2 ] 1/2   (4).  
         [0044]    The time that it takes for a signal to reach any given Distance can be found by dividing the Distance by the speed of sound (about 1129 feet/second at 68 degrees Fahrenheit). Since knowing the absolute time required for a signal to travel to each microphone is not necessary only the change in time from one microphone to the next is stored in step  420 .  
         [0045]    In another aspect of the invention, the beamforming parameters are calculated in the frequency domain instead of the time domain. As is well known in the art, Fourier theory (including both the Fourier Series and the Fourier Transform) relates the time domain and the frequency domain. Furthermore, it is often useful to use a Fast Fourier Transform (FFT) as an approximation to the Fourier Transformation. The effects of background noise can be easily reduced in the frequency domain by, for example, applying greater weights to some frequencies or implementing a bandwidth filter. In the frequency domain, the beamforming parameters represent the changes in phase from one microphone to the next, and not the changes in time.  
         [0046]    Referring back to FIG. 4, step  430  begins the process of camera tracking. In steps  430  sound is received by microphone array  150 . In step  440  each set of beaming parameters are applied to the received sound. In step  450  videoconferencing system  140  determines which set of beamforming parameters synchronizes the sound received by microphone array  150 . Then, in step  460 , the location of the audio source is identified by matching the synchronizing beamforming parameters with its associated potential source of sound.  
         [0047]    As an example of steps  430  through  460 , FIG. 6A shows graphs of a signal  610  received by three microphones in the time domain. The signal strength is shown along the Y-axis and time is represented along the X-axis. Signal  610  is received at separate times for each microphone due to varying microphone distances from the source of signal  610 . Specifically, an arbitrary component  620  within signal  610  peaks at microphone A at a time t a    630 . The same component  620  peaks at a time t b    634  for microphone B and at a time t c    638  for microphone C.  
         [0048]    [0048]FIG. 6B shows signal  610  modified by the set of beamforming parameters for a first potential source. The set of beamforming parameters are calculated during initialization (steps  410  and  420 ) and consist of two numbers (B b1 , B c1 ). A signal generated by a first potential source reaches microphone A at a time B b1  different than it reaches microphone B. B b1  can be either positive, negative or zero, depending upon whether the signal reaches microphone A before, after or at the same time that it reaches microphone B. Similarly, B c1  represents the time interval between microphone A and microphone C receiving the signal.  
         [0049]    In order to apply the beamforming parameters to the signal at microphone B, the entire signal is shifted by an amount equal to B b1 . Therefore, the arbitrary component  620  within signal  610  peaks at a time (t b −B b1 )  640 . Component  620  peaks at microphone C at a time (t c −B c1 )  645 . If the source of signal  610  were located at the first potential source, then both (t b −B b1 )  640  and (t c −B c1 )  650  would be equal to t a    630 .  
         [0050]    [0050]FIG. 6C shows signal  610  modified by the set of beamforming parameters (B b2 , B c2 ) for a second potential source. Since component  620  peaks at time t a    630 , which is equal to a time (t b −B b2 )  630  and a time (t c −B c2 )  630 , the source of the signal is located at the second potential source.  
         [0051]    There are many methods that may be used to determine whether the modified signals for each microphone match. For example, adding the modified signals from each microphone generates an aggregate signal for a potential source. The potential source that generates the loudest aggregate signal would be where the signal originated. When aggregate signal  660  in FIG. 6B (labeled “Sum 1 ”) is compared to aggregate signal  660  in FIG. 6C (labeled “Sum 2 ”), the peak sound intensity for aggregate signal  670  in FIG. 6C exceeds the peak sound intensity for aggregate signal  660  in FIG. 6B.  
         [0052]    In another aspect of the invention, calculations are performed in the frequency domain. As previously mentioned, a phase shift in the frequency domain is equivalent to a time shift in the time domain. Therefore, a method similar to that already described can be applied in the frequency domain.  
         [0053]    In the last step  470  a decision is made whether to continue camera tracking. Camera tracking can be performed only once, be performed only at certain intervals, be continuous, or only be performed only if certain conditions are met (e.g., when sound is received). Unnecessary camera movement may be avoided by waiting until a certain amount of time has elapsed before moving the camera to ensure a new person is speaking, thereby limiting erroneous tracking due to coughing or errant comments. Camera tracking can also be performed by using multiple cameras and only activating the camera that frames the speaker or by using a PTZ camera (mechanical or electronic).  
         [0054]    Although the invention has been described in its presently contemplated best mode, it is clear that it is susceptible to numerous modifications, modes of operation and embodiments, all within the ability and skill of those familiar with the art and without the exercise of further inventive activity. Accordingly, that which is intended to be protected by Letters Patents is set forth in the claims and includes all variations and modifications that fall within the spirit and scope of the invention.