Patent Grant
7826624
1. Field of the Invention
The present invention relates generally to the field of communication devices and, more specifically, to speakerphones.
2. Description of the Related Art
Speakerphones are used in many types of telephone calls, and particularly are used in conference calls where multiple people are located in a single room. A speakerphone may have a microphone to pick up voices of in-room participants, and, at least one speaker to audibly present voices from offsite participants. While speakerphones may allow several people to participate in a conference call on each end of the conference call, there are a number of problems associated with the use of speakerphones.
As the microphone and speaker age, their physical properties change, thus compromising the ability to perform high quality acoustic echo cancellation. Thus, there exists a need for a system and method capable of estimating descriptive parameters for the speaker and the microphone as they age.
Furthermore, noise sources such as fans, electrical appliances and air conditioning interfere with the ability to discern the voices of the conference participants. Thus, there exists a need for a system and method capable of “tuning in” on the voices of the conference participants and “tuning out” the noise sources.
In one set of embodiments, a system (e.g., a speakerphone or a videoconferencing system) may include a microphone, a speaker, memory and a processor. The memory may be configured to store program instructions and data. The processor is configured to read and execute the program instructions from the memory. The program instructions are executable by the processor to:
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
The input-output model of the speaker may be a nonlinear model, e.g., a Volterra series model.
The stimulus signal may be a noise signal, e.g., a burst of maximum-length-sequence noise.
Furthermore, the program instructions may be executable by the processor to:
The average transfer function may also be usable to perform said echo cancellation on said other input signals.
In another set of embodiments, a method for performing self calibration may involve:
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
The input-output model of the speaker may be a nonlinear model, e.g., a Volterra series model.
In yet another set of embodiments, a system (e.g., a speakerphone or a videoconferencing system) may include a microphone, a speaker, memory and a processor. The memory may be configured to store program instructions and data. The processor is configured to read and execute the program instructions from the memory. The program instructions are executable by the processor to:
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
The input-output model of the speaker is a nonlinear model, e.g., a Volterra series model.
Furthermore, the program instructions may be executable by the processor to:
The current transfer function is usable to perform said echo cancellation on said other input signals.
In yet another set of embodiments, a method for performing self calibration may involve:
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
Furthermore, the method may involve:
The current transfer function is also usable to perform said echo cancellation on said other input signals.
In yet another set of embodiments, a system may include a set of microphones, memory and a processor. The memory is configured to store program instructions and data. The processor is configured to read and execute the program instructions from the memory. The program instructions are executable by the processor to:
The program instructions are also executable by the processor to provide the resultant signal to a communication interface for transmission.
The set of microphones may be arranged in a circular array.
In yet another set of embodiments, a method for beam forming may involve:
The resultant signal may be provided to a communication interface for transmission (e.g., to a remote speakerphone).
The set of microphones may be arranged in a circular array.
In yet another set of embodiments, a system may include a set of microphones, memory and a processor. The memory is configured to store program instructions and data. The processor is configured to read and execute the program instructions from the memory. The program instructions are executable by the processor to:
The program instructions are executable by the processor to provide the resultant signal to a communication interface for transmission.
The set of microphones may be arranged in a circular array.
In yet another set of embodiments, a method for beam forming may involve:
The resultant signal may be provided to a communication interface for transmission (e.g., to a remote speakerphone).
The set of microphones are arranged in a circular array.
The following detailed description makes reference to the accompanying drawings, which are now briefly described.
While the invention is described herein by way of example for several embodiments and illustrative drawings, those skilled in the art will recognize that the invention is not limited to the embodiments or drawings described. It should be understood, that the drawings and detailed description thereto are not intended to limit the invention to the particular form disclosed, but on the contrary, the intention is to cover all modifications, equivalents and alternatives falling within the spirit and scope of the present invention as defined by the appended claims. The headings used herein are for organizational purposes only and are not meant to be used to limit the scope of the description or the claims. As used throughout this application, the word “may” is used in a permissive sense (i.e., meaning having the potential to), rather than the mandatory sense (i.e., meaning must). Similarly, the words “include”, “including”, and “includes” mean including, but not limited to.
Speakerphone Block Diagram
The processor 207 is configured to read program instructions which have been stored in memory 209 and to execute the program instructions to execute any of the various methods described herein.
Memory 209 may include any of various kinds of semiconductor memory or combinations thereof. For example, in one embodiment, memory 209 may include a combination of Flash ROM and DDR SDRAM.
The input subsystem may include a microphone 201 (e.g., an electret microphone), a microphone preamplifier 203 and an analog-to-digital (A/D) converter 205. The microphone 201 receives an acoustic signal A(t) from the environment and converts the acoustic signal into an electrical signal u(t). (The variable t denotes time.) The microphone preamplifier 203 amplifies the electrical signal u(t) to produce an amplified signal x(t). The A/D converter samples the amplified signal x(t) to generate digital input signal X(k). The digital input signal X(k) is provided to processor 207.
In some embodiments, the A/D converter may be configured to sample the amplified signal x(t) at least at the Nyquist rate for speech signals. In other embodiments, the A/D converter may be configured to sample the amplified signal x(t) at least at the Nyquist rate for audio signals.
Processor 207 may operate on the digital input signal X(k) to remove various sources of noise, and thus, generate a corrected microphone signal Z(k). The processor 207 may send the corrected microphone signal Z(k) to one or more remote devices (e.g., a remote speakerphone) through one or more of the set 211 of communication interfaces.
The set 211 of communication interfaces may include a number of interfaces for communicating with other devices (e.g., computers or other speakerphones) through well-known communication media. For example, in various embodiments, the set 211 includes a network interface (e.g., an Ethernet bridge), an ISDN interface, a PSTN interface, or, any combination of these interfaces.
The speakerphone 200 may be configured to communicate with other speakerphones over a network (e.g., an Internet Protocol based network) using the network interface. In one embodiment, the speakerphone 200 is configured so multiple speakerphones, including speakerphone 200, may be coupled together in a daisy chain configuration.
The output subsystem may include a digital-to-analog (D/A) converter 240, a power amplifier 250 and a speaker 225. The processor 207 may provide a digital output signal Y(k) to the D/A converter 240. The D/A converter 240 converts the digital output signal Y(k) to an analog signal y(t). The power amplifier 250 amplifies the analog signal y(t) to generate an amplified signal v(t). The amplified signal v(t) drives the speaker 225. The speaker 225 generates an acoustic output signal in response to the amplified signal v(t).
Processor 207 may receive a remote audio signal R(k) from a remote speakerphone through one of the communication interfaces and mix the remote audio signal R(k) with any locally generated signals (e.g., beeps or tones) in order to generate the digital output signal Y(k). Thus, the acoustic signal radiated by speaker 225 may be a replica of the acoustic signals (e.g., voice signals) produced by remote conference participants situated near the remote speakerphone.
In one alternative embodiment, the speakerphone may include circuitry external to the processor 207 to perform the mixing of the remote audio signal R(k) with any locally generated signals.
In general, the digital input signal X(k) represents a superposition of contributions due to:
Processor 207 may be configured to execute software including an automatic echo cancellation (AEC) module.
The AEC module attempts to estimate the sum C(k) of the contributions to the digital input signal X(k) due to the acoustic signal generated by the speaker and a number of its reflections, and, to subtract this sum C(k) from the digital input signal X(k) so that the corrected microphone signal Z(k) may be a higher quality representation of the acoustic signals generated by the conference participants.
In one set of embodiments, the AEC module may be configured to perform many (or all) of its operations in the frequency domain instead of in the time domain. Thus, the AEC module may:
An inverse Fourier transform may be performed on the spectrum Z(ω) to obtain the corrected microphone signal Z(k). As used herein, the “spectrum” of a signal is the Fourier transform (e.g., the FFT) of the signal.
In order to estimate the spectrum C(ω), the AEC module may operate on:
For example, the modeling information IM may include:
The parameters (d) may be (or may include) propagation delay times for the direct path transmission and a set of the reflected path transmissions between the output of speaker 225 and the input of microphone 201.
In some embodiments, the input-output model for the speaker may be (or may include) a nonlinear Volterra series model, e.g., a Volterra series model of the form:
where v(k) represents a discrete-time version of the speaker's input signal, where fs(k) represents a discrete-time version of the speaker's acoustic output signal, where Na, Nb and Mb are positive integers. For example, in one embodiment, Na=8, Nb=3 and Mb=2. Expression (1) has the form of a quadratic polynomial. Other embodiments using higher order polynomials are contemplated.
In alternative embodiments, the input-output model for the speaker is a transfer function (or equivalently, an impulse response).
The AEC module may compute an update for the parameters (d) based on the output spectrum Y(ω), the input spectrum X(ω), and at least a subset of the modeling information IM (possibly including previous values of the parameters (d)), and then, compute the compensation spectrum C(ω) using the output spectrum Y(ω) and the modeling information IM (including the updated values of the parameters (d)).
In those embodiments where the speaker input-output model is a nonlinear model (such as a Volterra series model), the AEC module may be able to converge more quickly and/or achieve greater accuracy in its estimation of the direct path and reflected path delay times because it will have access to a more accurate representation of the actual acoustic output of the speaker than in those embodiments where linear model (e.g., transfer function) is used to model the speaker.
In some embodiments, the AEC module may employ one or more computational algorithms that are well known in the field of echo cancellation.
The modeling information IM (or certain portions of the modeling information IM) may be initially determined by measurements performed at a testing facility prior to sale or distribution of the speakerphone 200. Furthermore, certain portions of the modeling information IM (e.g., those portions that are likely to change over time) may be repeatedly updated based on operations performed during the lifetime of the speakerphone 200.
In one embodiment, an update to the modeling information IM may be based on samples of the input signal X(k) and samples of the output signal Y(k) captured during periods of time when the speakerphone is not being used to conduct a conversation.
In another embodiment, an update to the modeling information IM may be based on samples of the input signal X(k) and samples of the output signal Y(k) captured while the speakerphone 200 is being used to conduct a conversation.
In yet another embodiment, both kinds of updates to the modeling information IM may be performed.
Updating Modeling Information Based on Offline Calibration Experiments
In one set of embodiments, the processor 207 may be programmed to update the modeling information IM during a period of time when the speakerphone 200 is not being used to conduct a conversation.
The processor 207 may wait for a period of relative silence in the acoustic environment. For example, if the average power in the input signal X(k) stays below a certain threshold for a certain minimum amount of time, the processor 207 may reckon that the acoustic environment is sufficiently silent for a calibration experiment. The calibration experiment may be performed as follows.
The processor 207 may output a known noise signal as the digital output signal Y(k). In some embodiments, the noise signal may be a burst of maximum-length-sequence noise, followed by a period of silence. For example, in one embodiment, the noise signal burst may be approximately 2-2.5 seconds long and the following silence period may be approximately 5 seconds long.
The processor 207 may capture a block BX of samples of the digital input signal X(k) in response to the noise signal transmission. The block BX may be sufficiently large to capture the response to the noise signal and a sufficient number of its reflections for a maximum expected room size.
The block BX of samples may be stored into a temporary buffer, e.g., a buffer which has been allocated in memory 209.
The processor 207 computes a Fast Fourier Transform (FFT) of the captured block BX of input signal samples X(k) and an FFT of a corresponding block BY of samples of the known noise signal Y(k), and computes an overall transfer function H(ω) for the current experiment according to the relation
H(ω)=FFT(BX)/FFT(BY), (2)
where ω denotes angular frequency. The processor may make special provisions to avoid division by zero.
The processor 207 may operate on the overall transfer function H(ω) to obtain a midrange sensitivity value s1 as follows.
The midrange sensitivity value s1 may be determined by computing an A-weighted average of the overall transfer function H(ω):
s1=SUM[H(ω)A(ω), ω ranging from zero to 2π]. (3)
In some embodiments, the weighting function A(ω) may be designed so as to have low amplitudes:
The diaphragm of an electret microphone is made of a flexible and electrically non-conductive material such as plastic (e.g., Mylar) as suggested in
As the microphone ages, the deposited charge slowly dissipates, resulting in a gradual loss of sensitivity over all frequencies. Furthermore, as the microphone ages material such as dust and smoke accumulates on the diaphragm, making it gradually less sensitive at high frequencies. The summation of the two effects implies that the amplitude of the microphone transfer function |Hmic(ω)| decreases at all frequencies, but decreases faster at high frequencies as suggested by
The speaker 225 includes a cone and a surround coupling the cone to a frame. The surround is made of a flexible material such as butyl rubber. As the surround ages it becomes more compliant, and thus, the speaker makes larger excursions from its quiescent position in response to the same current stimulus. This effect is more pronounced at lower frequencies and negligible at high frequencies. In addition, the longer excursions at low frequencies implies that the vibrational mechanism of the speaker is driven further into the nonlinear regime. Thus, if the microphone were ideal (i.e., did not change its properties over time), the amplitude of the overall transfer function H(ω) in expression (2) would increase at low frequencies and remain stable at high frequencies, as suggested by
The actual change to the overall transfer function H(ω) over time is due to a combination of affects including the speaker aging mechanism and the microphone aging mechanism just described.
In addition to the sensitivity value s1, the processor 207 may compute a lowpass sensitivity value s2 and a speaker related sensitivity s3 as follows. The lowpass sensitivity factor s2 may be determined by computing a lowpass weighted average of the overall transfer function H(ω):
s2=SUM[H(ω)L(ω), ω ranging from zero to 2π]. (4)
The lowpass weighting function L(ω) equals is equal (or approximately equal) to one at low frequencies and transitions towards zero in the neighborhood of a cutoff frequency. In one embodiment, the lowpass weighting function may smoothly transition to zero as suggested in
The processor 207 may compute the speaker-related sensitivity value s3 according to the expression:
s3=s2−s1.
The processor 207 may maintain sensitivity averages S1, S2 and S3 corresponding to the sensitivity values s1, s2 and s3 respectively. The average Si, i=1, 2, 3, represents the average of the sensitivity value si from past performances of the calibration experiment.
Furthermore, processor 207 may maintain averages Ai and Bij corresponding respectively to the coefficients ai and bij in the Volterra series speaker model. After computing sensitivity value s3, the processor may compute current estimates for the coefficients bij by performing an iterative search. Any of a wide variety of known search algorithms may be used to perform this iterative search.
In each iteration of the search, the processor may select values for the coefficients bij and then compute an estimated input signal XEST(k) based on:
The processor may compute the energy of the difference between the estimated input signal XEST(k) and the block BX of actually received input samples X(k). If the energy value is sufficiently small, the iterative search may terminate. If the energy value is not sufficiently small, the processor may select a new set of values for the coefficients bij, e.g., using knowledge of the energy values computed in the current iteration and one or more previous iterations.
The scaling of the linear terms in the modified Volterra series expression (5) by factor c serves to increase the probability of successful convergence of the bij.
After having obtained final values for the coefficients bij, the processor 207 may update the average values Bij according to the relations:
Bij←kijBij+(1−kij)bij, (6)
where the values kij are positive constants between zero and one.
In one embodiment, the processor 207 may update the averages Ai according to the relations:
Ai←giAi+(1−gi)(cAi), (7)
where the values gi are positive constants between zero and one.
In an alternative embodiment, the processor may compute current estimates for the Volterra series coefficients ai based on another iterative search, this time using the Volterra expression:
After having obtained final values for the coefficients ai, the processor may update the averages Ai according the relations:
Ai←giAi+(1−gi)ai. (8B)
The processor may then compute a current estimate Tmic of the microphone transfer function based on an iterative search, this time using the Volterra expression:
After having obtained a current estimate Tmic for the microphone transfer function, the processor may update an average microphone transfer function Hmic based on the relation:
Hmic(ω)←kmHmic(ω)+(1−km)Tmic(ω), (10)
where km is a positive constant between zero and one.
Furthermore, the processor may update the average sensitivity values S1, S2 and S3 based respectively on the currently computed sensitivities s1, s2, s3, according to the relations:
S1←h1S1+(1−h1)s1, (11)
S2←h2S2+(1−h2)s2, (12)
S3←h3S3+(1−h3)s3, (13)
where h1, h2, h3 are positive constants between zero and one.
In the discussion above, the average sensitivity values, the Volterra coefficient averages Ai and Bij and the average microphone transfer function Hmic are each updated according to an IIR filtering scheme. However, other filtering schemes are contemplated such as FIR filtering (at the expense of storing more past history data), various kinds of nonlinear filtering, etc.
In one set of embodiments, a system (e.g., a speakerphone or a videoconferencing system) may include a microphone, a speaker, memory and a processor, e.g., as illustrated in
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
The input-output model of the speaker may be a nonlinear model, e.g., a Volterra series model.
Furthermore, the program instructions may be executable by the processor to:
The average transfer function is also usable to perform said echo cancellation on said other input signals.
In another set of embodiments, as illustrated in
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
The input-output model of the speaker may be a nonlinear model, e.g., a Volterra series model.
Updating Modeling Information Based on Online Data Gathering
In one set of embodiments, the processor 207 may be programmed to update the modeling information IM during periods of time when the speakerphone 200 is being used to conduct a conversation.
Suppose speakerphone 200 is being used to conduct a conversation between one or more persons situated near the speakerphone 200 and one or more other persons situated near a remote speakerphone (or videoconferencing system). In this case, the processor 207 essentially sends out the remote audio signal R(k), provided by the remote speakerphone, as the digital output signal Y(k). It would probably be offensive to the local persons if the processor 207 interrupted the conversation to inject a noise transmission into the digital output stream Y(k) for the sake of self calibration. Thus, the processor 207 may perform its self calibration based on samples of the output signal Y(k) while it is “live”, i.e., carrying the audio information provided by the remote speakerphone. The self-calibration may be performed as follows.
The processor 207 may start storing samples of the output signal Y(k) into an first FIFO and storing samples of the input signal X(k) into a second FIFO, e.g., FIFOs allocated in memory 209. Furthermore, the processor may scan the samples of the output signal Y(k) to determine when the average power of the output signal Y(k) exceeds (or at least reaches) a certain power threshold. The processor 207 may terminate the storage of the output samples Y(k) into the first FIFO in response to this power condition being satisfied. However, the processor may delay the termination of storage of the input samples X(k) into the second FIFO to allow sufficient time for the capture of a full reverb tail corresponding to the output signal Y(k) for a maximum expected room size.
The processor 207 may then operate, as described above, on a block BY of output samples stored in the first FIFO and a block BX of input samples stored in the second FIFO to compute:
Because the block BX of received input sample is captured while the speakerphone 200 is being used to conduct a live conversation, the block BX is very likely to contain interference (from the point of view of the self calibration) due to the voices of persons in the environment of the microphone 201. Thus, in updating the average values with the respective current estimates, the processor may strongly weight the past history contribution, i.e., much more strongly than in those situations described above where the self-calibration is performed during periods of silence in the external environment.
In some embodiments, a system (e.g., a speakerphone or a videoconferencing system) may include a microphone, a speaker, memory and a processor, e.g., as illustrated in
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
The input-output model of the speaker is a nonlinear model, e.g., a Volterra series model.
Furthermore, the program instructions may be executable by the processor to:
The current transfer function is usable to perform said echo cancellation on said other input signals.
In one set of embodiments, as illustrated in
The parameter averages of the speaker input-output model are usable to perform echo cancellation on other input signals.
Furthermore, the method may involve:
The current transfer function is also usable to perform said echo cancellation on said other input signals.
Plurality of Microphones
In some embodiments, the speakerphone 200 may include NM input channels, where NM is two or greater. Each input channel ICj, j=1, 2, 3, . . . , NM may include a microphone Mj, a preamplifier PAj, and an A/D converter ADCj. The description given above of various embodiments in the context of one input channel naturally generalizes to NM input channels.
Let uj(t) denote the analog electrical signal captured by microphone Mj.
In one group of embodiments, the NM microphones may be arranged in a circular array with the speaker 225 situated at the center of the circle as suggested by the physical realization (viewed from above) illustrated in
Preamplifier PAj amplifies the difference signal rj(t) to generate an amplified signal xj(t). ADCj samples the amplified signal xj(t) to obtain a digital input signal Xj(k).
Processor 207 may receive the digital input signals Xj(k), j=1, 2, . . . , NM.
In one embodiment, NM equals 16. However, a wide variety of other values are contemplated for NM.
Hybrid Beamforming
In one set of embodiments, processor 207 may operate on the set of digital input signals Xj(k), j=1, 2, . . . , NM to generate a resultant signal D(k) that represents the output of a highly directional virtual microphone pointed in a target direction. The virtual microphone is configured to be much more sensitive in an angular neighborhood of the target direction than outside this angular neighborhood. The virtual microphone allows the speakerphone to “tune in” on any acoustic sources in the angular neighborhood and to “tune out” (or suppress) acoustic sources outside the angular neighborhood.
According to one methodology, the processor 207 may generate the resultant signal D(k) by:
Each of the virtual beams B(i), i=1, 2, . . . , NB has an associated frequency range
R(i)=[ci,di]
and operates on a corresponding subset Si of the input spectra Xj(f), j=1, 2, . . . , NM. (To say that A is a subset of B does not exclude the possibility that subset A may equal set B.) The processor 207 may window each of the spectra of the subset Si with a window function Wi corresponding to the frequency range R(i) to obtain windowed spectra, and, operate on the windowed spectra with the beam B(i) to obtain spectrum V(i). The window function Wi may equal one inside the range R(i) and the value zero outside the range R(i). Alternatively, the window function Wi may smoothly transition to zero in neighborhoods of boundary frequencies ci and di.
The union of the ranges R(1), R(2), . . . , R(NB) may cover the range of audio frequencies, or, at least the range of frequencies occurring in speech.
The ranges R(1), R(2), . . . , R(NB) includes a first subset of ranges that are above a certain frequency fTR and a second subset of ranges that are below the frequency fTR. For example, in one embodiment, the frequency fTR may be approximately 550 Hz.
Each of the virtual beams B(i) that corresponds to a frequency range R(i) below the frequency fTR may be a beam of order L(i) formed from L(i)+1 of the input spectra Xj(f), j=1, 2, . . . , NM, where L(i) is an integer greater than or equal to one. The L(i)+1 spectra may correspond to L(i)+1 microphones of the circular array that are aligned (or approximately aligned) in the target direction.
Furthermore, each of the virtual beams B(i) that corresponds to a frequency range R(i) above the frequency fTR may have the form of a delay-and-sum beam. The delay-and-sum parameters of the virtual beam B(i) may be designed by beam forming design software. The beam forming design software may be conventional software known to those skilled in the art of beam forming. For example, the beam forming design software may be software that is available as part of MATLAB®.
The beam forming design software may be directed to design an optimal delay-and-sum beam for beam B(i) at some frequency (e.g., the midpoint frequency) in the frequency range R(i) given the geometry of the circular array and beam constraints such as passband ripple δP, stopband ripple δS, passband edges θP1 and θP2, first stopband edge θS1 and second stopband edge θS2 as suggested by
The beams corresponding to frequency ranges above the frequency fTR are referred to herein as “high end” beams. The beams corresponding to frequency ranges below the frequency fTR are referred to herein as “low end” beams. The virtual beams B(1), B(2), . . . , B(NB) may include one or more low end beams and one or more high end beams.
In some embodiments, the beam constraints may be the same for all high end beams B(i). The passband edges θP1 and θP2 may be selected so as to define an angular sector of size 360/NM degrees (or approximately this size). The passband may be centered on the target direction θT.
The delay-and-sum parameters for each high end beam and the parameters for each low end beam may be designed at a laboratory facility and stored into memory 209 prior to operation of the speakerphone 200. Since the microphone array is symmetric with respect to rotation through any multiple of 360/NM degrees, the set of parameters designed for one target direction may be used for any of the NM target directions given by k(360/NM), k=0, 1, 2, . . . , NM−1.
In one embodiment,
In another embodiment, the virtual beams B(1), B(2), . . . , B(NB) may include a set of low end beams of first order.
In yet another embodiment, the virtual beams B(1), B(2), . . . , B(NB) may include a set of low end beams of third order.
In one embodiment, the low order beams may include:
For example, f1 may equal approximately 250 Hz.
In some embodiments, a system (e.g., a speakerphone or a videoconferencing system) may include a set of microphones, memory and a processor, e.g., as suggested in
The program instructions are also executable by the processor to provide the resultant signal to a communication interface for transmission.
The set of microphones may be arranged in a circular array.
In another set of embodiments, as illustrated in
The resultant signal may be provided to a communication interface for transmission (e.g., to a remote speakerphone).
The set of microphones may be arranged in a circular array.
The high end beams may be designed using beam forming design software. Each of the high end beams may be designed subject to the same (or similar) beam constraints. For example, each of the high end beams may be constrained to have the same pass band width (i.e., main lobe width).
In yet another set of embodiments, a system may include a set of microphones, memory and a processor, e.g., as suggested in
The program instructions are executable by the processor to provide the resultant signal to a communication interface for transmission.
The set of microphones may be arranged in a circular array.
In yet another set of embodiments, as illustrated in
The resultant signal may be provided to a communication interface for transmission (e.g., to a remote speakerphone).
The set of microphones are arranged in a circular array.
The high end beams may be designed using beam forming design software. Each of the high end beams may be designed subject to the same (or similar) beam constraints. For example, each of the high end beams may be constrained to have the same pass band width (i.e., main lobe width).
Various embodiments may further include receiving, sending or storing program instructions and/or data implemented in accordance with the foregoing description upon a computer-accessible medium. Generally speaking, a computer-accessible medium may include storage media or memory media such as magnetic or optical media, e.g., disk or CD-ROM, volatile or non-volatile media such as RAM (e.g. SDRAM, DDR SDRAM, RDRAM, SRAM, etc.), ROM, etc. as well as transmission media or signals such as electrical, electromagnetic, or digital signals, conveyed via a communication medium such as network and/or a wireless link.
The various methods as illustrated in the Figures and described herein represent exemplary embodiments of methods. The methods may be implemented in software, hardware, or a combination thereof. The order of method may be changed, and various elements may be added, reordered, combined, omitted, modified, etc.
Various modifications and changes may be made as would be obvious to a person skilled in the art having the benefit of this disclosure. It is intended that the invention embrace all such modifications and changes and, accordingly, the above description to be regarded in an illustrative rather than a restrictive sense.
This application claims the benefit of priority to U.S. Provisional Application No. 60/619,303, filed on Oct. 15, 2004, entitled “Speakerphone”, invented by William V. Oxford, Michael L. Kenoyer and Simon Dudley, which is hereby incorporated by reference in its entirety. This application claims the benefit of priority to U.S. Provisional Application No. 60/634,315, filed on Dec. 8, 2004, entitled “Speakerphone”, invented by William V. Oxford, Michael L. Kenoyer and Simon Dudley, which is hereby incorporated by reference in its entirety.
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