An array of microphones on an aircraft that determines a location and steers the transducer beam to that location

Abstract

Audio content is controllably directed to or from an aircraft occupant (such as a pilot or a passenger) using a microphone array of at least two transducers, the microphone array being positioned on an aircraft to pick up the speech of the aircraft occupant. An electronically controllable transducer array coupled to an onboard aircraft audio system (such as an avionics communication system or an in-flight entertainment system) provides the audio content to the aircraft occupant. A signal processor coupled to the microphone array processes the speech sensed by the transducer using an arrival time algorithm to determine the direction of origin of the speech. Then, a controller circuit provides a pointing direction control signal to a controllable beamforming speaker array or microphone array based on the direction of origin of the speech, so that the controllable transducer array directs the audio content in the direction of the sensed speech, and thereby directs the audio content to or from the aircraft occupant.

Description

Technical Field

[0001] The present disclosure relates generally to audio sound systems for aircraft, and more particularly to an electrically operated speaker or microphone array that projects or receives sound in the direction of a user by detecting the sound of the user's voice. Background Art

[0002] This section provides background information related to the present disclosure which is not necessarily prior art.

[0003] The interior of a modern business aircraft defines an elongated tubular acoustic space. The pilot and co-pilot sit on the flight deck at the nose of the aircraft, where the flight controls and avionics are located. The rest of the occupied space is usually equipped with seating areas, in-flight entertainment systems, sleeping areas, lavatories, etc.

[0004] Traditionally, headsets are typically used at each crew seat position. Pilots use an earmuff-style headset with an attached boom microphone, which is used to communicate with air traffic control via the avionics communications system. Passengers are typically issued a less refined headset, which is used primarily to enjoy the in-flight entertainment system. In some cases, speakers placed above or near the passenger may be used instead of a headset.

[0005] The advantage of headsets is that they can dampen ambient aircraft aerodynamic and engine noise, but this comes at the expense of comfort, which some find awkward or uncomfortable when worn for long periods of time. Summary of the invention

[0006] The disclosed system and method captures arrival direction information from the spoken utterance of a pilot or aircraft passenger, and uses the information to steer the beam pattern of a transducer array such as a speaker array or a microphone array. In the case of steering a speaker array, the arrival direction information is used to focus or steer the output of the speaker array in the direction of the spoken utterance, thereby allowing the pilot or passenger to hear more clearly. In the case of steering a microphone array, the arrival direction information is used to train the receiving pattern of the microphone array in the direction of the spoken utterance, thereby improving the speech clarity of the pilot or passenger-which may be particularly useful in a noisy aircraft environment. The arrival direction information from the spoken utterance can also be used to control cabin control systems, such as lighting, curtain control, in-flight entertainment system control, etc. In such applications, the arrival direction information allows the controlled system to apply control to a specific location within the aircraft, such as the location where the pilot or passenger is seated. Thus, for example, a voice command to lower an automatic curtain will be applied to the curtain near the location where the spoken utterance originated.

[0007] The disclosed system and method can eliminate the need for pilots and passengers to wear uncomfortable headsets. In one embodiment, the system uses an electrically steerable speaker array deployed at each seat or potential listening position. In one embodiment, the electrically steerable speaker array uses phased array technology to focus enhanced sound pressure levels in the direction of the steered beam pointing, while providing greatly reduced sound pressure levels at angles outside the beam. In one embodiment, the speaker array is implemented using ultrasonic transducers driven by an ultrasonic (inaudible to the human ear) carrier wave that is modulated with audio program content.

[0008] To steer the speaker array to the location occupied by the user, the system employs clusters of spaced-apart microphone transducers deployed throughout the aircraft near each seat position or other commonly occupied location. When an occupant speaks, his or her voice is detected by the nearby transducer clusters, and the system uses a signal processor programmed with a direction-of-arrival algorithm to locate the position of the speaking occupant. This position is then fed to the steerable speaker array to point the steering beam in the direction of the speaking user.

[0009] Due to the relatively narrow beamwidth of each speaker array, the occupants at which the speakers are now pointed will hear the audio program content, while occupants elsewhere in the aircraft will not hear it. In properly equipped aircraft, this "personal speaker" effect can be enhanced by equipping the aircraft with sound absorbing materials on interior wall surfaces, partition surfaces, and carpeted floors.

[0010] The disclosed systems and methods are well suited for use on a flight deck, allowing the pilot to hear communications with air traffic control, avionics warning systems, and private intercom communications, etc., without having to wear a headset and without disturbing passengers. Similarly, passengers can selectively enjoy the in-flight entertainment content of their choice without having to wear a headset and without disturbing other passengers who may be listening to a different entertainment program or trying to sleep.

[0011] According to one aspect, the disclosed method controllably directs audio content to an aircraft occupant by receiving sounds within an aircraft using a microphone array of at least two transducers. The received sounds are processed to extract human speech, and a direction of arrival analysis is performed on the extracted human speech to determine a speech origin direction. The speech origin direction is used to electronically steer a speaker array toward the speech origin direction. Using the speaker array, the audio content is then directed from the aircraft audio system to the aircraft occupant from whom the extracted human speech originated.

[0012] According to another aspect, the disclosed system controllably directs audio content to an aircraft occupant. A microphone array of at least two transducers is positioned on an aircraft to pick up speech of an aircraft occupant. An electronically controllable speaker array coupled to an onboard aircraft audio system provides the audio content to the aircraft occupant. A signal processor is coupled to the microphone array to process speech sensed by the at least two transducers to determine a speech origin direction of the sensed speech. A controller circuit then provides a pointing direction control signal to the controllable speaker array based on the speech origin direction to cause the controllable speaker array to direct the audio content in the direction of the sensed speech and thereby direct the audio content to the aircraft occupant.

[0013] According to another aspect, the disclosed system controls an aircraft cabin system of a type that provides functionality at a plurality of different selectable locations. A microphone array of at least two transducers receives aircraft sounds. These received sounds are processed to extract human speech. Direction of arrival analysis is then performed on the extracted human speech to determine a speech origin direction. Using the speech origin direction, the cabin system is caused to provide functionality at at least one selected location based on the speech origin direction. BRIEF DESCRIPTION OF THE DRAWINGS

[0014] The drawings described herein are for illustrative purposes only of selected embodiments and not all possible implementations. The particular selection of drawings is not intended to limit the scope of the present disclosure.

[0015] Figure 1 is a plan view of a portion of occupied space within an exemplary commercial aircraft showing some locations where a microphone array and a steerable speaker array may be deployed;

[0016] Figure 2 yes Figure 1 A plan view of a flight deck of an aircraft showing in more detail an exemplary arrangement of microphone arrays and steerable speaker arrays for the pilot and co-pilot;

[0017] Figure 3a and Figure 3b Diagrammatically illustrating a sequence by which a pilot's speech is detected and used to steer a speaker array in the direction of the pilot;

[0018] Figure 4 is a block diagram showing how microphone array and speaker array systems can be combined with signal processing to steer a beamforming speaker array or a beamforming microphone array pointing direction based on detected speech;

[0019] Figure 5 is a waveform diagram showing how different arrival times of the same spoken signal may be measured;

[0020] Figure 6 is a diagram showing how the geometric origin of a wavefront of a spoken sound may arrive at a microphone transducer pair from different viewing angles;

[0021] Figure 7 is a diagram showing how phasing (inserted time delay) of a pair of loudspeaker transducers within a loudspeaker array can affect beam pointing direction through constructive and destructive interference;

[0022] Figure 8 is a flow chart for understanding how a programmed processor may implement an audio directional system; and

[0023] Fig. 9 is a block diagram for understanding the GCC-PHAT maximum likelihood estimation technique for direction of arrival analysis. DETAILED DESCRIPTION

[0024] refer to Figure 1 , the interior plan cabin layout of an exemplary business aircraft is shown at 10. A flight deck 12 occupies the front portion of the interior space within the nose of the aircraft, while the remainder of the occupied interior space is used for passenger areas 14, including areas for flight crew. The walls and ceiling of the interior space may be treated with sound absorbing materials, and the floor may also be provided with carpet, which also acts as a sound absorber. Different parts of the cabin may be separated by partitions to define different functional areas. For example, in Figure 1 In the embodiment, a partition is provided at 16 to define a more private sleeping area 18 .

[0025] Depending on the intended use of the aircraft, various passenger seats 20 are deployed throughout the cabin, as well as work spaces and dining tables 22. Individual in-flight entertainment system monitors, such as drop-down, ceiling-mounted monitors, are provided for each passenger seat and sofa bed 26. In the cabin layout shown, a large-screen in-flight entertainment system monitor 24 is positioned opposite the sofa 28 to allow several people to view the monitor simultaneously.

[0026] Deployed on suitable structures adjacent to each occupied area within the passenger area, galley, lavatory and flight deck are miniature microphone arrays 30 and speaker arrays 32 (in Figure 1 Not shown in Figure 2). As will be further explained, the microphone array is positioned to pick up speech (e.g., human speech) from passengers, pilots, and other crew members. Signal processing circuitry analyzes these received speech to determine the location where the speech originated. The beam steering system then uses this information to electronically steer the beams of a nearby speaker array. The same information will also be used to steer the beamforming microphone array to improve the intelligibility of passengers' speech in noisy environments. The system is therefore able to provide audio content directly to or from each occupant's location. Figure 2 The relative arrangement of the microphone array 30 and the speaker array 32 is not necessarily drawn to scale, and the actual mounting locations may vary in practice, as the actual arrangement of these components may depend on where a suitable mounting structure is located on the flight deck.

[0027] refer to Figure 2 The flight console 12 is separated from the passenger area by a partition 16. In the example shown, three pilot seats 34 are provided for the pilot, co-pilot and auxiliary pilot. Mounted in appropriate locations on the instrument panel are microphone arrays 30 (two such arrays are shown, one for the pilot and one for the co-pilot). Speaker arrays 32 are also appropriately mounted (two such arrays are shown, one for the pilot and one for the co-pilot). Figure 2 Schematically shown are electronic components for implementing the audio directional system 36. Also schematically shown at 38 is an avionics communication system that provides radio communications between the pilot and air traffic control and other services, including intercom communications with the crew, and also provides audible warning and alert information. The avionics communication system provides audio content through a speaker array 32 positioned in front of the pilot.

[0028] Figure 3a and Figure 3b shows how the audio directional system works. Figure 3a As shown in FIG. 4 , pilot 40a speaks a verbal message that is transmitted as speech 42, which is received by microphone array 30a. Electronic audio directional system 36 processes the received speech to determine the direction from which the speech originated, and then uses the originating direction to electronically steer the acoustic output of speaker array 32 (located closest to pilot 40a) in the direction of pilot 40a. Figure 3b, having now been steered to direct its acoustic energy toward the pilot, the speaker array 32 transmits subsequent audio content from the avionics communication system 38 to the pilot as an acoustic energy beam 44. The audio directing system 36 may be configured to perform a sample and hold function whereby, having learned the position of a person from his or her speech, the system will maintain the speaker array 32 trained on the learned position unless subsequent speech dictates that the beam direction should be adjusted.

[0029] refer to Figure 4 , the audio directional system 36 is shown in more detail. In the steered speaker mode, the system is designed to direct audio content from the speaker array 32 to an occupant of the aircraft, such as a pilot or passenger 40, after sensing the direction or position of the occupant by determining the origin direction of the utterance 42 spoken by the occupant. In the focused microphone mode, the system is also designed to focus the beamforming microphone array 30 (including multiple microphone transducers 46) in the direction of the occupant after sensing the direction or position of the occupant. If desired, the application of steered speakers and focused microphones can be used together.

[0030] As previously discussed, speech 42 is picked up by microphone array 30, which includes a plurality of individually addressable (independently monitored) microphone transducers 46 arranged in predetermined groups of known positions and spacings. Speaker array 32 also includes a plurality of individually addressable speaker transducers 50 of known positions and spacings.

[0031] The individual microphone transducers 46 are each coupled to a multi-channel preamplifier 52, which amplifies the corresponding output of each microphone transducer 46 to a suitable level to drive a multi-channel analog-to-digital (A to D) converter 54. Since the audio directional system relies on arrival time detection, the preamplifier 52 and the analog-to-digital converter circuit 54 should be configured to introduce equal phase delays (or time delays) (if any) on all channels. This will ensure that the measurement of the relative arrival time is accurately determined to ensure that the calculated time difference is accurate. This can be achieved in part by carefully selecting components to match the impedance of the microphone transducers and minimize capacitance or inductance differences between channels, and by ensuring that the electrical signal path lengths of all channels are the same. The system can also be implemented using microphone transducers with digital outputs that do not require analog-to-digital converters.

[0032] The multi-channel analog-to-digital converter 54 provides a digital representation of the output of each microphone transducer to a signal processor 56, which uses a direction of arrival algorithm to determine the location or position of the source of the speech. The signal processor 56 is programmed to correlate and compare the sounds received by the transducers of the microphone array 30 to determine the time when the same speech arrives at each transducer. Based on these determined times, the time difference dt is calculated.

[0033] like Figure 5 As shown diagrammatically in FIG, speech 42a arrives at first transducer 46a before speech 42b arrives at transducer 46b. The signal processor performs correlation on the sounds received by each transducer to identify that speech 42a and 42b represent the same speech event, but are received at different times. The time difference dt is calculated by the processor by subtracting the arrival times of the corresponding received speech signals. The location or position of the source of the speech is then calculated based on the calculated time difference dt.

[0034] Figure 6 shows how the calculated difference dt in the time of arrival at transducers 46a and 46b can be used to determine not only the general direction from which the utterance came, but also the precise location or position of the source of the utterance 42 by triangulation. Figure 6 In the 1980s, the words originated in 42 places and were Figure 6 The spherical wavefront depicted by the extended arc in is propagating acoustically. As shown, the wavefront passes through and is received by transducer 46a at 48 before passing through and being received by transducer 46b. The spacing L between transducers 46a and 46b is fixed. Therefore, the line-of-sight distance x from the speech source 42 to transducer 46a is shorter than the line-of-sight distance y by an amount equal to the additional distance the wavefront propagates to 46b during the calculated delay time dt. In other words, knowing the calculated or measured time delay dt and the speed of sound c in the aircraft cabin, the distances from the speech source 42 to the respective transducers 46a and 46b can be related as follows:

[0035] y=x+dt×c

[0036] Having calculated the distances x and y and knowing the fixed spacing L and the known positions of the microphone transducers 46a and 46b, the direction and position of the speech source 42 can be determined. If desired, an angular measure can be used to represent the position of the speech source. Figure 6 In FIG. 4 , such angular measurements are shown as θ1 and θ2 measured relative to a line perpendicular to the plane in which transducers 46a and 46b lie. It will be appreciated that Figure 5 and Figure 6It is intended to simply conceptually illustrate how the source of speech is related to the arrival time at multiple transducers whose positions and spacings are known. There are a variety of computational techniques that can be used to determine the direction of arrival and the spatial position or location of the source of speech. One embodiment of doing this is to program a processor to identify the source of speech by maximizing the generalized cross-correlation phase transform (GCC-PHAT) function, as described below.

[0037] Focus Mic Mode

[0038] return Figure 4 , the processor 56 uses the calculated speech source calculation to provide the arrival direction instruction to the microphone array beamforming delay adjustment processor 57. This inserts a time delay or phase change into the selected signal in the input signal from the analog-to-digital converter 54 to cause the microphone array to operate as a steered beamforming array, which directs or focuses the microphone array pickup pattern to enhance the sound pickup in the direction of the speech source. The output of the microphone array beamforming delay adjustment processor 57 is fed as an audio input source at 24 and 38. In this way, the audio signal picked up by the microphone array 30 will prefer the sound pickup in the direction where the beam has been actively focused. As described above, the focused microphone mode can be used simultaneously with the steered speaker mode.

[0039] Controlling speaker modes

[0040] return Figure 4 , the processor 56 uses the calculated speech source calculation to provide instructions to the beam steering circuit or signal processing operation 60. The actual steering of the acoustic beam is performed without the need for physical reorientation of the speaker transducer 50 or the microphone array 30. Instead, the beam is steered electronically by introducing timing or phasing differences in the audio signal fed to each individual transducer. These phase-shifted or time-shifted audio signals are provided to the individual speaker transducers 50 of the speaker array 32 through appropriate driver circuits 72 and power amplifiers 74. Note that the phase-shifted or time-shifted audio signals are transmitted to the speaker transducers of the speaker array in parallel. It should be noted that the driver circuit 72 and the power amplifier 74 introduce equal phase delays (or time delays) (if any) on all channels. This will ensure that appropriate constructive and destructive interference will occur in order to properly steer the beam. This can be achieved in part by carefully selecting components to match the impedance of the speaker transducers and minimize capacitance or inductance differences between channels, and by ensuring that the electrical signal path lengths of all channels are the same.

[0041] The beam steering circuit or process 60 essentially transmits the audio content from the avionics communication system 38 (for the pilot) or from the in-flight entertainment system 24 (for the passengers) after inserting the calculated time delay for each channel to achieve the beam steering. The audio input from the avionics communication system 38 and the in-flight entertainment system 24 may be converted by analog-to-digital conversion in the audio processing circuit 62 if not already in the digital domain.

[0042] Modulated Ultrasonic Beam Embodiment

[0043] If desired, a more directional (narrower) beam 44 may be achieved using an ultrasonic speaker system, which may be implemented as part of the audio processing circuit 62, such as Figure 4 64 in the figure. In this embodiment, the audio input from the avionics communication system 38 and / or the in-flight entertainment system 24 is provided to the modulator circuit 68 after being processed by the beam steering adjustment 60, and the modulator circuit 68 modulates the audio program content onto the ultrasonic carrier provided by the ultrasonic oscillator 66. One such modulator circuit is used to feed each of the driver 72 circuits so that each transducer transmits the ultrasonic signal with an appropriate time delay (phase adjustment). In this way, the speaker array generates an ultrasonic beam with an ultrasonic carrier at a frequency inaudible to the human ear. The system audio is modulated on the ultrasonic carrier and thus propagates in the direction of the beam with the carrier. When the beam passes through air molecules on the way to the listener's ears, the modulated wave becomes naturally demodulated by interacting with the air molecules. Due to this natural demodulation, the listener is able to hear the demodulated audio signal, which sounds like it comes from a conventional speaker.

[0044] Control speaker beam direction

[0045] The signal processor 56 uses the arrival time delays between the microphone transducer signals to electronically steer the beam 44 in the direction of the person whose speech is to be received. This is achieved by inserting a controlled delay.

[0046] To better understand how beam steering is achieved, please refer to Figure 7 , which shows two loudspeaker transducers 50 arranged in a linear array. When the two loudspeakers are coherently fed with the same audio signal, the sound waves emitted from each loudspeaker are in phase, and the sound will appear to be emitted from the plane of the loudspeakers (i.e., from a direction perpendicular to the horizontal axis (e.g., Figure 7 The direction shown in ) is directly generated.

[0047] However, when one of the loudspeakers is fed with a signal delayed by a time increment dt, constructive and destructive interference between the corresponding wavefronts of the two loudspeakers will produce the loudest collective sound in an angled direction, no longer perpendicular to the horizontal axis but at an angle θ, such as Figure 7 By knowing the wavelength of the audio frequency, the angular direction can be calculated trigonometrically. The frequency (f) and wavelength λ are related to the speed of sound c according to the following equation:

[0048] f=c / λ

[0049] In order to Figure 7 The beam is steered in the direction (angle θ) shown in the figure, coming from the left speaker ( Figure 7 The signal from the left speaker ( Figure 7 The signal from the right speaker (center) must travel an additional distance d so that its wavefront coincides with that from the right speaker ( Figure 7 The wavefronts in the middle are in phase. This delay dt can be calculated for a given angle θ using the following trigonometric relationship:

[0050] dt=s sin(θ) / c

[0051] where s is the speaker spacing and c is the speed of sound at ambient temperature.

[0052] Programming Processor Implementation

[0053] Figure 8 One exemplary embodiment of the steps a processor is programmed to perform is shown, beginning at 100. Although the processor is depicted as a series of steps in sequence, these operations may be performed in parallel if desired.

[0054] The processor uses a memory 102, which is configured to store calibration data 104, which is used to give the processor knowledge of the positions of the microphone transducers and the speaker transducers. Alternatively, if necessary, the transducer positions can be hard-coded in firmware or software. The memory 102 also provides data storage 106 for storing the last used (previous) speech source position. The processor uses this stored position to calculate the speaker pointing direction, and if the speaker moves, the stored value is updated as needed-effectively implementing a sample and hold function. The memory 102 is also used to define a plurality of predefined audio channels 108, including a set of channels for each microphone transducer and a set of channels for each speaker transducer. These are digital audio channels, and digital audio from the microphone and from the avionics communication system 38 or the in-flight entertainment system is stored in buffer memory locations of appropriate size, and the processor can operate on these buffer memory locations to manipulate the data.

[0055] The processor receives input from a microphone at 110 and applies a pre-filter at 112. As discussed below, the pre-filter can be used to suppress or reject sounds received by the microphone that do not have human voice quality. At 114, the processor applies an arrival time algorithm or function to the pre-filtered (human voice utterance) data to find the utterance source location. In one embodiment, the GCC-PHAT function is used.

[0056] The processor tests at 116 whether the utterance source location is substantially different from a value previously stored in the data store 106. If so, the processor proceeds to step 118 and writes the new utterance source location in the data store 106. If not, the processor reverts to start 100.

[0057] Based on the speech source position values ​​stored in the data store 106, the processor calculates a new speaker array pointing direction at step 120. Figure 6 The processor then uses the new pointing direction at 122 to calculate the time delay parameters that will cause the speaker beam to point in the new pointing direction. Figure 7 The processor uses these calculated time delay parameters to insert time delays in the audio channels of each speaker when applicable at step 124. This is accomplished by operating on the buffered digital audio data in the audio channels to reposition certain data for delayed playback (relative to other audio channels). The time-delayed audio data is provided as audio content through multiple signal channels for each speaker.

[0058] Generalized Cross-Correlation Phase Transform (GCC-PHAT)

[0059] refer to Figure 4 In one embodiment, the processor 56 is programmed to perform arrival time detection by maximizing the generalized cross-correlation phase transform (GCC-PHAT) function 58 to determine the source (location and / or direction) of the speech. This is done by comparing the signals received from two spatially separated microphone transducers by applying the signals to a cross-correlator with an adjustable time component, which establishes a time delay of one signal relative to the other. The two signals are then multiplied and integrated for various different delay combinations by programmatically or algorithmically adjusting the time component until a maximum peak is found. The time component delay that achieves this maximum peak corresponds to the maximum likelihood estimate of the arrival time delay between the two signals.

[0060] Fig. 9The basic maximum likelihood estimation principle is shown. Two microphone signals x1 and x2 are applied to a filter 200 having filter functions H1 and H2. One of the two signal paths is delayed (as at 202). The signals are then multiplied at 204, integrated at 206 and squared at 208 to produce a composite signal that is analyzed by a peak detector 210. The two microphone signals are processed in this manner while adjusting the delay applied at 202 until the maximum peak is found at the peak detector. This process implements a maximum likelihood estimator that includes a pair of receiver prefilters followed by a cross-correlator. The time argument at which the correlator reaches a maximum is the delay estimate.

[0061] Different filter functions or transforms may be used for filter 200. In the described embodiment, a phase transform (PHAT) has been used. Other filter alternatives may be substituted. If desired, the GCC-PHAT function may be implemented using the Matlab gccphat(sig, refsig) function.

[0062] Pre-filtering

[0063] The GCC-PHAT function used by the signal processor 56 is fairly robust in the presence of background noise, assuming that the background noise is relatively uncorrelated. However, to enhance speech detection and avoid false triggering, the signal processor 56 may also include one or more audio pre-filters to process the microphone transducer data prior to the arrival time estimation (GCC-PHAT). Such pre-filtering may include frequency band filtering, in which microphone transducer signals outside the normal human speech frequency range are suppressed. To further improve scalability, different paired microphone arrays may be used to derive more reliable arrival direction estimates.

[0064] Such pre-filtering may also include Mel frequency processing, which more closely approximates the response of the human auditory system than linearly spaced frequency bands. In addition, pre-filtering may also include time windowing, whereby the input data stream from the microphone transducer is sampled into time windows corresponding to the natural duration of human speech, thereby allowing the processor to suppress arrival time estimation processing for signals that may include the same frequencies as human speech but do not exist on a time frame comparable to human speech.

[0065] In more advanced systems, pre-filtering may also include speech recognition processing, whereby the processor tests whether the received signal generates a high enough likelihood score from the speech recognizer algorithm to conclude that the signal is human speech. In this case, the recognizer does not need to decode the utterance into text, but only determines that the received sound is likely human speech. Combinations of these and other similar filters may also be used.

[0066] Cabin system control mode

[0067] In addition to manipulating the speaker mode and the focus microphone mode, the arrival direction signal processor 56 ( Figure 4 ) can also interface with the onboard cabin management system 75 to provide focused position control of these systems.

[0068] There are many passenger convenience systems in modern business aircraft, including in-flight entertainment systems, lighting controls, automatic window shade controls, etc. These systems will typically provide selectable functionality at multiple different discrete locations throughout the aircraft. For example, individual task lighting is provided above each seating position, in addition to cabin lighting to provide ambient light and illuminate the aisles. This lighting may be equipped with dimmable switch controls and selectable color temperature control. In such lighting systems, passengers expect to be able to control the lighting for their location without affecting the lighting for other passengers. This is also true for automatic window shades and in-flight entertainment systems.

[0069] In order to provide each passenger with individually focused control of their space within the aircraft, the arrival direction processor 56 provides location information about each passenger to the cabin management system 75. Thus, when a passenger wishes to control the cabin management system, the system knows where the passenger is located, thereby limiting the control commands to the appropriate location of the passenger. If desired, the cabin management system 75 can be operated using voice commands. Due to the arrival direction information from the processor 56, the control system does not need to specify the passenger location in advance in a programmatic manner.

[0070] Although at least one exemplary embodiment has been proposed in the foregoing detailed description, it should be understood that there are a large number of variations. It should also be understood that the exemplary embodiment or multiple exemplary embodiments are only examples and are not intended to limit the scope, applicability or configuration of the present invention in any way. On the contrary, the foregoing detailed description will provide a convenient roadmap for those skilled in the art to implement the exemplary embodiments as considered herein. It should be understood that various changes can be made in the functions and arrangements of the elements described in the exemplary embodiments without departing from the scope of the present invention as set forth in the appended claims.

Claims

1. A method of controllably directing audio content to or from an aircraft occupant, comprising: A microphone array using at least two transducers receives sounds within the aircraft; Processing the received sound to extract human speech; performing an arrival direction analysis on the extracted human utterances to determine the utterance origin direction; electronically steer a transducer array toward the utterance origin direction using the utterance origin direction, wherein the transducer array is a beamforming transducer array; Audio content is directed to or from the aircraft occupant from which the extracted human speech originated using the transducer array.

2. The method of claim 1 , wherein the transducer array is a beamforming speaker array, and further comprising using the speaker array to direct audio content from an aircraft audio system to the aircraft occupant from whom the extracted human speech originated.

3. The method of claim 1, wherein the transducer array is a beamforming microphone array that is electronically steered to improve speech intelligibility in the direction from which the utterance originates.

4. The method of claim 1, wherein processing the received sound to extract human speech is performed using a process selected from the group consisting of frequency band processing, mel frequency processing, time window processing, speech recognition processing, and combinations thereof.

5. The method of claim 1 , wherein the direction of arrival analysis is performed using a process selected from the group consisting of a steered beamformer process, a subspace-based process, a time delay estimation process, a maximization process, a cross-correlation process, a generalized cross-correlation phase transform process, and combinations thereof.

6. The method of claim 2, wherein the speaker array comprises a plurality of spaced-apart speaker transducers, and wherein the speaker array is operated by individually providing the audio content from the aircraft audio system to each speaker transducer at a controlled time based on the utterance origination direction.

7. The method of claim 2, wherein the speaker array comprises a plurality of spaced-apart ultrasonic transducers, and wherein the speaker array is operated by providing an ultrasonic carrier wave modulated to carry the audio content to each ultrasonic transducer, and The ultrasonic carrier wave is individually provided to each speaker transducer with a controlled phase delay based on the utterance origin direction to produce an ultrasonic wave beam steered in the utterance origin direction.

8. The method according to claim 2, further comprising: deploying the at least two transducers within a passenger cabin of the aircraft to monitor speech from passengers; The speaker array is deployed within the passenger cabin of the aircraft and coupled to an in-flight entertainment system to direct audio content from the in-flight entertainment system to the passengers.

9. The method according to claim 2, further comprising: deploying the at least two transducers within a flight deck of the aircraft to monitor speech from a pilot; The speaker array is deployed within the flight deck of the aircraft and coupled to an avionics communication system to direct audio content from the avionics communication system to the pilot.

10. A method of controlling an aircraft cabin system, the aircraft cabin system providing functionality at a plurality of different selectable locations, the method comprising: A microphone array using at least two transducers receives sounds within the aircraft; Processing the received sound to extract human speech; performing an arrival direction analysis on the extracted human utterances to determine the utterance origin direction; Using the utterance origin direction, a transducer array is electronically steered toward the utterance origin direction and a cabin system provides a function at at least one selected location based on the utterance origin direction, wherein the transducer array is a beamforming transducer array.

11. A system for controllably directing audio content to or from an aircraft occupant, comprising: a microphone array of at least two transducers positioned on the aircraft to pick up speech of an occupant of the aircraft, wherein the microphone array is a beamforming microphone array; an electronically steerable speaker array coupled to an onboard aircraft audio system to provide audio content to an occupant of said aircraft, wherein said speaker array is a beamforming speaker array; a signal processor coupled to the microphone array to process speech sensed by the at least two transducers to determine a speech origination direction of the sensed speech; a controller circuit that provides a pointing direction control signal to a steerable speaker array based on the utterance originating direction to cause the steerable speaker array to direct the audio content in the direction of the sensed utterance and thereby direct the audio content toward the aircraft occupant; The same or a separate controller circuit provides a directional control signal to a controllable microphone array based on the speech origin direction, so that the controllable microphone array improves the speech clarity in the speech origin direction and reduces interference or noise from other directions.

12. The system of claim 11 wherein the aircraft audio system comprises an avionics communications system and wherein a microphone array and an electronically controllable speaker array are positioned within a flight deck of the aircraft to monitor speech from a pilot and direct audio content from the avionics communications system to or from the pilot.

13. The system of claim 11, wherein the aircraft audio system is an in-flight entertainment system, and wherein a microphone array and an electronically controllable speaker array are positioned within a passenger cabin of the aircraft to monitor speech from passengers and direct audio content from the in-flight entertainment system to the passengers.

14. The system of claim 11, wherein the signal processor includes a filter to suppress signals generated by sounds sensed by the at least two transducers that lack characteristics of human speech.

15. The system of claim 14, wherein the filter suppresses signals lacking substantial energy at frequencies associated with human speech.

16. The system of claim 11, wherein at least one of the controller circuits includes a sample and hold circuit that maintains the pointing direction of the speaker array or the microphone array as the last sensed utterance until the next sensed utterance is processed.

17. The system of claim 11, wherein the signal processor pre-processes the audio signals sensed by the at least two transducers to suppress the effects of sounds not spoken by the aircraft occupant.

18. The system of claim 11, wherein the signal processor pre-processes the audio signals sensed by the at least two transducers using a process selected from the group consisting of frequency band processing, mel frequency processing, time window processing, speech recognition processing, and combinations thereof.

19. A system according to claim 11, wherein the signal processor is programmed to apply an arrival direction algorithm to the speech sensed by the at least two transducers, and the arrival direction algorithm is selected from the group consisting of a steered beamformer process, a subspace-based process, a time delay estimation process, a maximization process, a cross-correlation process, a generalized cross-correlation phase transform process and combinations thereof.

20. The system of claim 11, wherein the speaker array comprises a plurality of spaced-apart ultrasonic transducers that transmit the audio content on modulated ultrasonic beams.

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