Abstract: A digital audio signal processing system and method transforms two-channel stereo time-domain data into the frequency domain. Vector operations are performed upon the frequency-domain data by which signal components unique to one of the input channels are routed to one of the output channels, signal components unique to the other of the input channels are routed to another of the output channels, and signal components common to both channels are routed to a third and optionally to a fourth output channel. The frequency-domain output channels are then transformed back into the time-domain, forming an equivalent number of channels of output audio data. The vector operations are performed in a manner that preserves the overall information content of the input data.

Abstract: The system of the present invention includes, briefly, a point source speaker system, comprising a processor which produces a left minus right (L?R) audio signal, a right plus left (R+L) and a right minus left (R?L) audio signal; three speakers each for audibly transmitting one of the L?R, R+L and R?L audio signals; and a point source speaker enclosure for housing the three speakers in a single enclosure.

Abstract: The invention relates to a method for automated tuning of a sound system, the sound system comprising delay lines, equalizing filters, and at least two loudspeakers, the method comprising the steps of reproducing a useful sound signal through the loudspeakers, measuring sound pressure values at least one location, providing a target transfer function for tuning the delay lines and the equalizing filters of the sound system, the target transfer function representing a desired transfer characteristics of the sound system, adjusting the delay of the delay lines, and adjusting amplitude responses of the equalizing filters such, that the actual transfer characteristics of the sound system approximates the target function.

Abstract: A sound reproduction system comprising a digital audio signal input (1), a digital audio signal processor (2, DSP) and a digital audio signal output (3) wherein the digital signal processor (2, DSP) comprises a high pass (HP) filter (21) with a high pass frequency (f), an amplifier (22) for a signal filtered by the HP filter, and a low pass (LP) filter (23) with a low pass frequency (f) for filtering the signal after amplification by the amplifier (22) and for providing an output signal, and the digital processor comprises an establisher (24, 25) for establishing the high pass frequency or the low pass frequency and a matcher (26) for matching the high pass frequency and low pass frequency of the high pass filter and low pass filter respectively to each other.

Abstract: A speaker system having a first speaker, a second speaker, and a housing having a first mounting surface to which the first speaker is mounted and a second mounting surface to which the second speaker is mounted. The first mounting surface and the second mounting surface are substantially planar and abut one another to form an included angle of between 120 degrees and 140 degrees. By so arranging the mounting surfaces, the sound produced by each speaker is allowed to mix to provide a full sound.

Abstract: In a frequency characteristic adjustment apparatus incorporated in an audio system, a frequency characteristic of an audio signal is made to agree with a target frequency characteristic, thus providing an equalizing function. In the apparatus, by a frequency characteristic controller, a measurement signal is detected as a sound-collected signal that received by way of speakers/microphone, and the sound-collected signal is divided into a signal component falling into a one fixed-level band and one or more signal components falling into one or more variable-level bands. A relative level of the signal component in each variable-level band is estimated on the basis of a level of the signal component in the fixed-level band in the target frequency characteristic. Based on the estimated relative levels, the levels at equalizers are set, so that the levels of signal components of the signal are adjusted by the equalizers channel by channel.

Abstract: To improve upon a method of adjusting at least one filter parameter such as the mid-frequency of at least one filter unit provided with a playback system for audio signals, this filter unit being provided in particular for at least one digital equalizer assigned to the playback system, as well as an associated playback system so as to achieve automatic optimization of an acoustically correct bass level boosting, [a] first the acoustic frequency response is determined, [b] then a cutoff frequency below which the sound pressure falls below a sound pressure threshold level based on the average sound pressure is determined, and [c] next the filter parameter(s) of the filter unit is/are adjusted automatically, so that the mid-frequency of the filter unit is just above the cutoff frequency thus determined.

Abstract: A sound device is provided with an input terminal for electronic signal input; a low pass filter adapted to filter electronic signals input to the input terminal and to output low frequency range electronic signals, a switch arrangement adapted to switch between full range mode and sub woofer mode, a power amplifier adapted to amplify (1) electronic signals input from the input terminal when the switch arrangement selects full range mode, and (2) electronic signals output from the low pass filter when the switch arrangement selects sub woofer mode, and a speaker adapted to be operated by electronic signals amplified by the power amplifier.

Abstract: The present invention relates to a modular loudspeaker, primarily intended to be used as a studio monitor and comprising a broadband module for radiating a first frequency range with a lower limiting frequency sufficiently low to provide a fully acceptable reproduction in many situations and furthermore provided with a high frequency radiating unit (and possibly also a mid frequency radiating unit), the directional characteristics of which relative to the cabinet of the module can be controlled. The modular loudspeaker according to the invention furthermore comprises one or more low frequency module(s) for radiating a second frequency range which may at least partially overlap said first frequency range for either increasing the acoustical output at the lower end of the frequency range radiated by the broadband module or for extending the lower limiting frequency of the complete modular loudspeaker below the lower limiting frequency of the broadband module.

Abstract: A method and apparatus to enhance one or more low-frequency components and one or more medium-frequency components of an audio signal. The method includes performing filtering on the input audio signal using a plurality of band-pass filters, generating a plurality of harmonic-frequency signals using a plurality of audio signals resulting from the performing filtering operation, and mixing the plurality of harmonic-frequency signals with the input audio signal.

Abstract: A new approach has been developed to combine mid-range and high frequency sound into the throat of a horn designed for sound reinforcement. An acoustic low pass filter element is interposed between the lower frequency passage and the higher frequency passage, so that a smooth combination of the two frequency bands is achieved at the entrance to the horn bell. Thus, each frequency band has nearly identical dispersion, and the two sources have equal delay.

Abstract: An audio processor (202) receives a non-priority audio signal (302) and a priority audio signal (304). The priority audio signal occupies a frequency band (408). The audio processor filters (320) the non-priority audio signal by suppressing frequency content in the same frequency region occupied by the priority signal, creating a filtered non-priority signal (412). The filtered non-priority signal and the priority signal are combined (328) and played over an audio transducer (110).

Abstract: A method of communicating with an ingestible capsule includes detecting the location of the ingestible capsule, focusing a multi-sensor acoustic array on the ingestible capsule, and communicating an acoustic information exchange with the ingestible capsule via the multi-sensor acoustic array. The ingestible capsule includes a sensor that receives a stimulus inside the gastrointestinal tract of an animal, a bidirectional acoustic information communications module that transmits an acoustic information signal containing information from the sensor, and an acoustically transmissive encapsulation that substantially encloses the sensor and communications module, wherein the acoustically transmissive encapsulation is of ingestible size. The multi-sensor array includes a plurality of acoustic transducers that receive an acoustic signal from a movable device, and a plurality of delays, wherein each delay is coupled to a corresponding acoustic transducer.

Abstract: A two-channel to three-channel audio converter that separates the lower frequency output signals from one of the left channel and the right channel is provided. Respective filters are used to separate the lower frequency outputs and the middle to high frequency outputs from the audio output signals of the computer device. An equalizer balances the powers of the left channel and the right channel outputs, so that the mismatches in the resistance may be avoided.

Abstract: A two-channel to three-channel audio converter that separates the lower frequency output signals from one of the left channel and the right channel is provided. Respective filters are used to separate the lower frequency outputs and the middle to high frequency outputs from the audio output signals of the computer device. An equalizer balances the powers of the left channel and the right channel outputs, so that the mismatches in the resistance may be avoided. A wireless receiver is integrated into the two-channel to three-channel audio converter to facilitate the multimedia applications.

Abstract: The invention provides a multiple audio channel bass management method for an audio system comprising multiple processing units, wherein each processing unit receives a corresponding channel signal and outputs to a speaker, the method comprising: high pass filtering a first channel signal to drive a first speaker and low pass filtering the first channel signal to generate a first low frequency signal if the first speaker is not a full range speaker; combining a first bass information signal and the first channel signal to drive the first speaker if the first speaker is a full range speaker, wherein the first bass information signal is derived from the low frequency signal generated by other processing unit.

Abstract: A set of filters is configured to distribute input signals representing a single perceptual axis to first and second physically separate arrays of loudspeakers comprising at least first and second transducers, such that the arrays of loudspeakers will create an array pattern corresponding to the input signals when the input signals are between a first frequency and a second frequency.

Abstract: A loudspeaker is provided for receiving an incoming electrical signal and transmitting an acoustical signal. The loudspeaker may include a power amplifier that has an input and an output, where the input receives the incoming electrical signal. The loudspeaker may also include two or more passive filters, such as low-pass, band-pass, and/or high-pass filters, which are coupled to the output of the power amplifier. The passive filters may also be coupled to one or more speaker drivers. The arrangement of passive filters and speaker drivers may have a single input that has a combined input impedance. The output of the amplifier may have an output impedance. The output impedance may be between about 25% and about 400% of the combined input impedance. The power amplifier may include a current-feedback amplifier that is configured to maintain the desired output impedance.

Abstract: Separating a source in a stereo signal having a left channel and a right channel includes transforming the signal into a short-time transform domain; classifying portions of the signals having similar panning coefficients; segregating a selected one of the classified portions of the signals corresponding to the source; and reconstructing the source from the selected portions of the signals.

Abstract: A computer audio system includes an audio codec and a tone controller. The audio codec is operably coupled to receive audio information, which includes tone control settings, PCM digital audio inputs and PCM digital audio outputs. In addition, the audio codec may receive audio information as analog input signals via a line-in, a CD input, or an auxiliary input. Based on the audio information, the audio codec provides a first stereo output, a second stereo output and a monotone audio output. The tone controller is operably coupled to the audio codec and includes a low pass filter, a high pass filter, a band pass filter, and a summing module. The low pass filter is operably coupled to filter the monotone audio output and isolates bass components of the audio signal being processed. By further coupling a volume control module to the low pass filter, the bass component of the audio signal being processed may be varied.

Abstract: Separating a source in a stereo signal having a left channel and a right channel includes transforming the signal into a short-time transform domain; computing a short-time similarity measure between the left channel and the right channel; classifying portions of the signals having similar panning coefficients according to the short-time similarity measure; segregating a selected one of the classified portions of the signals corresponding to the source; and reconstructing the source from the selected portions of the signals.

Abstract: An audio limiting circuit capable of satisfying frequency dependent limits and time domain constraints, is disclosed. In one illustrative embodiment, an input node receives an unattenuated input signal and a system modeling filter predicts the amount, if any, by which the sound pressure level that would be generated by an acoustic transducer in response to the unattenuated input signal, would exceed one or more predetermined limits. In that embodiment, an energy detector separates the excess predicted sound pressure level into one or more frequency bands and calculates the average acoustic energy associated with each band. A gain logic block determines an attenuation factor based on whether one or more of the predetermined limits has been exceeded and the attenuation factor values are smoothed to minimize abrupt changes to the unattenuated input signal. A delay buffer delays the unattenuated input signal values.

Abstract: The present invention uses multivoice coil drivers in conjunction with the small satellite speakers of multichannel surround sound systems to achieve an overall linear response between 100 Hz to above 15 KHz. The multivoice coil drivers are designed to respond only to frequencies that fall between conventional upper base and lower midrange crossover frequencies which typically extend between 75 Hz and 225 Hz. The invention relies on the characteristics of human hearing which limits the listener's ability to localize the source of sound approximately to the wavelength of the sound being heard. The system may be incorporated in existing front and rear center speakers or separate enclosures to supplement the overall surround sound system.

Abstract: A sound reproduction device of the present invention includes: a first loudspeaker; a second loudspeaker; a first loudspeaker drive section for driving the first and second loudspeakers, the first loudspeaker drive section drives the first and second loudspeakers at substantially the same phase in a first frequency band, and the first loudspeaker drive section drives the first and second loudspeakers at substantially inverse phase and substantially different amplitude levels in a second frequency band higher than the first frequency band.

Abstract: Conventional analog loudspeakers have a limited dynamic range as compared to the available dynamic range of digital recordings. Digital recordings use up to 24 bits and this implies a dynamic range of 141 dB. Digital loudspeakers, involving 2N single bit devices (with N=24, this number is 1.7×107) have been proposed. The present improvement is the provision of at least one loudspeaker, a plurality of analog drivers and the audio input supplied to a control processor which in turn drives one or more of the plurality of independent analog drivers. The number of drivers in operation at any one time is determined by the amplitude of the input audio signal to the control processor.

Abstract: A reconfigurable speaker system for producing sound in response to at least one of right and left channel audio signals generated by an audio source. The speaker system can be configured by the user to simultaneously reproduce right and left channel audio information from a single speaker, or the speaker system can be configured to reproduce right and left channel audio information independently from a pair of speakers. The speaker system includes a dual voice coil woofer and a pair of independent tweeters. A switch device is provided in circuit with the woofer and pair of tweeters so that in one position of the switch device, the dual voice coil woofer is configured to produce sound in response to both of the right and left channel audio signals, and each tweeter in the tweeter pair is configured to produce sound in response to a respective one of the right and left channel audio signals.

Abstract: An improvement to inventor's prior art loudspeaker crossover filter invention is described. A substantially flat network input-impedance characteristic across the audible frequency range is provided by the addition of at least one series connected constant resistance network at the input terminals of the crossover filter system. By relaxing certain infinite slope filter parameters, more uniform acoustic polar response of the loudspeaker system is obtained. Specifically, infinite slope methods are used for the upper (higher frequency) band-edge of low-pass and band-pass filters, and optionally, for high-pass filters. Infinite slope characteristics, however, are relaxed for the lower frequency slope of band-pass filters and, optionally, for high-pass filters. In addition, the crossover network uses fewer components than infinite slope crossovers of the prior art.

Abstract: A midrange loudspeaker for operation in conjunction with low-frequency and high-frequency loudspeaker modules in a theater sound system, having a reduced depth for deployment in limited space. The midrange module is configured with a plurality of drivers and a waveguide unit that provides uniform sound coverage throughout a theater auditorium with substantially seamless crossovers at 250 Hz and 1.5 kHz and with the vertical beam-width held substantially constant by an electrical filter network.

Abstract: A system and method for adjusting the frequency response of a main speaker system used in conjunction with a subwoofer by modifying the input signal in response to user-adjustable settings to compensate for main-speaker effects characteristics in order to produce a desirable blend of the subwoofer sonic output and the main-speaker sonic output.

Abstract: A system for distributing a source voltage from a signal source to a number of drivers or loudspeakers is disclosed. The system includes at least one autotransformer for connection to the signal source, and a number of drivers electrically connected to the autotransformer. The autotransformers distribute the source voltage across each of the drivers. The autotransformers produce an output voltage across each of the drivers, such that the sum of the output voltages is substantially equal to the source voltage multiplied by the number of drivers.

Abstract: A computing device allows selective output of audio signals within a frequency band. The computing device has a main system having an audio signal generator generating audio signals, a main audio outputting unit attached to the main system and outputting the generated audio signals, an auxiliary system connected to the main system and receiving the audio signals generated by the audio signal generating unit, and an auxiliary audio outputting unit attached to the auxiliary system and outputting the audio signals received in the auxiliary system A switching unit controls the audio signals to pass through a plurality of selectable filters filtering the audio signals within respective frequency bands and to supply the selected filtered audio signals to the main audio outputting unit, and to the auxiliary audio outputting unit, depending upon whether the auxiliary system is connected to the main system.

Abstract: A sound reproduction system has pairs of sound emitters that subtend different angles ?, the span angle, at the listener position. The pairs of sound emitters are arranged to be excited by different frequency bands of the signal output from an inverse filter means (Hh, Hl). The operational span-frequency range of the pairs of sound emitters is determined by an equation (I) where the transducer span ? is the angle subtended at the listener by a pair of transducers, where O<n<2. c0: is the speed of sound, and ?r: is the equivalent distance between the ears. The sound emitters may be discrete speaker units, the different pairs of units being positioned at different span angles, or they be constituted by area portions of an extended transducer (FIG. 10). When discrete speaker units are employed a cross-over filter (FIG. 28) is used to provide drive signals in the different frequency bands to the different speaker pairs.

Abstract: In accordance with an embodiment of the present invention, a dynamics processor includes a non-linear automatic gain control (AGC) responsive to an input audio signal comprised of a plurality of frequency components, each frequency component having associated therewith an amplitude, said non-linear AGC adaptive to develop a modified gain audio signal. A multiband cross-over device is responsive to the modified gain audio signal and is adaptive to generate ‘n’ number of signals, each of said ‘n’ signals having an amplitude and further having a unique frequency band associated therewith.

Abstract: A computer audio system includes an audio codec and a tone controller. The audio codec is operably coupled to receive audio information, which includes tone control settings, PCM digital audio inputs and PCM digital audio outputs. In addition, the audio codec may receive audio information as analog input signals via a line-in, a CD input, or an auxiliary input. Based on the audio information, the audio codec provides a first stereo output, a second stereo output and a monotone audio output. The tone controller is operably coupled to the audio codec and includes a low pass filter, a high pass filter, a band pass filter, and a summing module. The low pass filter is operably coupled to filter the monotone audio output and isolates bass components of the audio signal being processed. By further coupling a volume control module to the low pass filter, the bass component of the audio signal being processed may be varied.

Abstract: A filter system including a low pass filter having a response which rolls off towards a crossover frequency and a high pass filter having a complementary response which rolls off towards the crossover frequency. The responses are arranged such that the combined response of the filters is substantially constant in amplitude at least in the region of the crossover frequency. The response of the low pass filter is defined by a low pass complex transfer function having a first numerator and a first denominator. The response of the high pass filter is defined by a high pass complex transfer function having a second numerator and a second denominator. The desired response is obtained when the second denominator is substantially the same as the first denominator and the sum of the first and second numerators has substantially the same squared modulus as the first or second denominator.

Abstract: A loudspeaker system, which has the ability to create a homogeneous sound over large distances, comprising at least one line-source of acoustical radiation, each comprising three or more essentially identical speaker-units arranged adjacent to each other at a center-to-center spacing D1, and at least one elongated high frequency transducer arranged in parallel with said line source(s), said elongated high frequency transducer(s) having an essentially continuous radiating surface along the axis of elongation. Where the crossover frequency FCR and the distance D1, for said line-source(s), is set according to new design criteria, taking into account the frequency response from said speaker units in said line-sources and the optimal performance for a line-source. A method is also included, by which a homogeneous sound over large distances can be accomplished.

Abstract: In a sound signal playback machine and method thereof, predetermined high frequency components are extracted from sound signals SL to SC of a main channel by high frequency pass filters 21 to 25, and these high frequency components are respectively played back by speakers. At the same time, predetermined low frequency components are extracted from sound signals SL to SC of the main channel by low frequency pass filters. These low frequency components and sound signal SLFE exclusively used for the low frequency band are added to each other, and the thus obtained addition signal is played back by the speaker 66. In this case, the degree of the low frequency pass filters is set higher than the degree of the high frequency pass filters, and the high frequency component is delayed.

Abstract: The invention relates to a filter circuit for processing an audio signal, comprising a first branch in which the audio signal is guided in an essentially unchanged state to an output summation stage, and a second branch comprising several series-connected filter stages. The audio signal is inputted into a first filter stage of the series-connected filter stages and guided via said stages to an output summation stage. The second branch comprises several secondary branches via which the audio signal is directly guided to further filter stages of the series-connected filter stages. At least one of the filter stages is adjustable and adjusting one filter stage has an impact on the function of another filter stage.

Abstract: An audio frequency distribution and adjusting circuit includes a variable resistor network having a first variable resistance and a second variable resistance coupled in series between a first resistor network terminal and a second resistor network terminal. The variable resistor network also includes a wiper contact with the first variable resistance and the second variable resistance for adjusting the total resistance between the resistive network terminals. The variable resistor network terminals are coupled in parallel with a low-pass filter, which is coupled to a low-frequency driver. The resistor network wiper contact is electrically coupled to the input of the high-pass filter, which is coupled to a high-frequency driver.

Abstract: A sub woofer system is constituted of a real time digital signal processing part 5 that includes an A/D converter 5a, a low pass filter block 5b, a delay block 5c, and a D/A converter block 5d, an analog power amplifier 6 and a speaker 7. The length of delay time of the delay block 5c is set so that the length of group delay time of the digital signal processing part 5 may equal an integral multiple of the length of time corresponding to one wavelength of the crossover point frequency between the digital signal processing part 5 and a main speaker 4. The signal processing is performed under the conditions that have been set as such. In this case, a construction is made up wherein the length of a group delay time produced in the digital type real time digital signal processing part can be set to a given value. By this construction, it is possible to provide a sub woofer system which can improve the phase interference between the main speaker and the sub woofer system at the crossover point between the two.

Abstract: Comprising:

Abstract: The system of the present invention includes, briefly, a point source speaker system, comprising a processor which produces a left minus right (L−R) audio signal, a right plus left (R+L) and a right minus left (R−L) audio signal; three speakers each for audibly transmitting one of the L−R, R+L and R−L audio signals; and a point source speaker enclosure for housing the three speakers in a single enclosure.

Abstract: Sound processing systems have been developed that create a surround effect without quality degradation experienced by known sound processing systems in non-optimum listening environments. The sound processing systems may include matrix decoding systems that manipulate input signals prior to converting them into a number of output signals so that the output signals are a function of a greater number of input signals. These sound processing systems may also or alternately include a base management system that from the input signals preserves the low frequency components of the input signals in separate channels. Both the matrix decoding systems and base management systems may also produce additional signals. Further, the matrix decoding and base management systems may be implemented separately or jointly in vehicular sound systems.

Abstract: A system (10) for providing an integer number N of filters. The system comprises an input (Di) for receiving a digital audio signal and an output (Do) for providing a filtered audio signal. The system also comprises circuitry (16) for storing at least a first set of fixed filter coefficients and circuitry for storing estimation data. The system also comprises circuitry (14) for estimating a number of sets of estimated filter coefficients in response to the estimation data and the fixed filter coefficients. The system also comprises circuitry (14) for applying a transfer function to the digital audio signal and in response for providing the filtered audio signal. The circuitry for applying the transfer function applies a set of filter coefficients selected from the first set of fixed filter coefficients and the sets of estimated filter coefficients. Also, the transfer function is selected from a transfer function set consisting of a high pass filter transfer function and a low pass filter transfer function.

Abstract: A speaker system having a first speaker, a second speaker, and a housing having a first mounting surface to which the first speaker is mounted and a second mounting surface to which the second speaker is mounted. The first mounting surface and the second mounting surface are substantially planar and abut one another to form an included angle of between 120 degrees and 140 degrees. By so arranging the mounting surfaces, the sound produced by each speaker is allowed to mix to provide a full sound.

Abstract: This invention unifies a set of statistical signal processing, neuromorphic systems, and microelectronic implementation techniques for blind separation and recovery of mixed signals. A set of architectures, frameworks, algorithms, and devices for separating, discriminating, and recovering original signal sources by processing a set of received mixtures and functions of said signals are described. The adaptation inherent in the referenced architectures, frameworks, algorithms, and devices is based on processing of the received, measured, recorded or otherwise stored signals or functions thereof. There are multiple criteria that can be used alone or in conjunction with other criteria for achieving the separation and recovery of the original signal content from the signal mixtures. The composition adopts both discrete-time and continuous-time formulations with a view towards implementations in the digital as well as the analog domains of microelectronic circuits.

Abstract: A system and method for adjusting a subwoofer sonic output in response to known main speaker characteristics in order to produce a desirable blending of sound from the combined subwoofer-main speaker output.

Abstract: A new approach has been developed to combine mid-range and high frequency sound into the throat of a horn designed for sound reinforcement. An acoustic low pass filter element is interposed between the lower frequency passage and the higher frequency passage, so that a smooth combination of the two frequency bands is achieved at the entrance to the horn bell. Thus, each frequency band has nearly identical dispersion, and the two sources have equal delay.

Abstract: An audio reproduction apparatus includes a primary audio amplifier having an output side for outputting audio signals, and a plurality of auxiliary audio amplifiers coupled electrically in series to the primary audio amplifier. Each of the auxiliary audio amplifiers has an input side for receiving the audio signals from the primary audio amplifier, a signal current generating device coupled electrically across the input side such that the audio signals pass through the signal current generating device, a current inducing device disposed adjacent to the signal current generating device for generating induced audio signals when the audio signals pass through the signal current generating device, an auxiliary amplifier coupled electrically to the current inducing device for amplifying and outputting the induced audio signals generated thereby, and a loudspeaker coupled electrically to the auxiliary amplifier for reproducing the induced audio signals amplified by the auxiliary amplifier.

Abstract: An adaptive speaker compensation system and method, such as for use in a multimedia computer, stores speaker response filter coefficients for each speaker and adaptively compensates received audio for non-linear speaker characteristics based on the stored speaker response filter coefficients. The speaker response filter coefficients preferably represent an inverse response of a speaker response curve for each speaker in the audio system. Preferably a library memory containing prestored speaker characteristic data, such as the speaker response filter coefficients, is selectively accessed by the adaptive speaker compensation system to download the speaker response filter coefficients based on identification of a speaker type and channel for which the speaker is being used.