Abstract: The present invention provides for methods and systems for digitally processing audio signals by adjusting the gain of a signal a first time, using a digital processing device located between a radio head unit and a speaker. The methods and systems may filter the adjusted signal with a first low shelf filter, compress the filtered signal with a first compressor, process the signal with a graphic equalizer, compress the processed signal with a second compressor and adjust the gain of the compressed signal a second time. The methods and systems may also filter the signal received from the first low shelf filter with a first high shelf filter prior to compressing the filtered signal with the first compressor. The signal may be filtered with a second low shelf filter prior to processing the signal with the graphic equalizer. Some embodiments filter the signal with a second high shelf filter after the signal is filtered with the second low shelf filter.

Abstract: Integrated processing of audio/video signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated system includes a main player that captures and processes signals digitally, a dummy display, and a dummy speaker. The dummy display may only have a display panel and a panel driver. The dummy speaker may only have a driving unit and no crossover logic. The main player may have a PC architecture and process all signals digitally for outputting signals tailored for the display device and the individual driving units of the dummy speaker. The integrated system may also provide dynamic signal adjustments based on the surrounding environment. The main player may include a storage device and can process media content stored therein to produce supplemental information to provide an optimal audiovisual experience. This supplemental information can be shared among users over a network connection.

Abstract: An audio crossover system and method is disclosed. An audio system includes two driver circuits, one for each of two audio frequency ranges, e.g., high and low frequency ranges. The driver circuits are designed to provide a combined frequency response curve that has a pronounced midrange attenuation dip, in contrast to prior art designs that attempt to provide a flat response over all frequency ranges.

Abstract: An audio device includes a first audio path with a loudspeaker for reproducing an audio signal, and a second audio path. The second audio path includes in series a band-pass filter for filtering an audio signal, a detector for detecting the amplitude of the band-pass filtered audio signal, a multiplier for multiplying a periodic signal by the amplitude of the band-pass filtered audio signal, and a vibration device for reproducing the multiplied periodic signal. The frequency of the periodic signal is substantially equal to the resonance frequency of the vibration device.

Abstract: The invention is a multi-channel loudspeaker system that provides a compact loudspeaker configuration and filter design methodology that operates in the digital signal processing domain. Further, the loudspeaker system can be designed to include drivers of various physical dimensions and can achieve prescribed constant directivity over a large area in both the vertical and horizontal planes.

Abstract: The invention is a multi-channel loudspeaker system that provides a compact loudspeaker configuration and filter design methodology that operates in the digital signal processing domain. Further, the loudspeaker system can be designed as a multi-way loudspeaker system comprised of a symmetric arrangement of loudspeaker drivers in a two-dimensional plane and can achieve high-quality sound, constant directivity over a large area in both the vertical and horizontal planes and can be used in connection with stereo loudspeaker systems, multi-channel home entertainment systems and public address systems.

Abstract: The present invention provides for methods and systems for digitally processing an audio signal. Specifically, the present invention provides for a headliner speaker system that is configured to digitally process an audio signal in a manner such that studio-quality sound that can be reproduced.

Abstract: Signals on respective RL, FL, C, FR, RR channels are divided into high frequency signals and low frequency signals by HPFs and LPFs respectively. The low frequency signals on the RL, FL, and C channels are superposed and output from a left-side woofer 21-1, while the low frequency signals on the RR, FR, and C channels are superposed and output from a right-side woofer 21-2. A predetermined directivity is given to the high frequency signals on respective channels by directivity controlling portions 17-1 to 17-5 respectively, and resultant signals are output from respective speaker units 20-1 to 20-n of an array speaker to generate virtual sound sources by the reflection from wall surfaces. A crossover frequency f2 of the rear channels (RL, RR) is set higher than a crossover frequency f1 of the front channels (FL, FR), and the signals on the rear channels are shaped into a narrow beam to generate a high-quality surround sound field.

Abstract: A method for performance measurement and optimization of sound systems using electroacoustic measurements and a sliding band integration curve. Nearfield and spatially and temporally averaged broadband farfield responses are measured, averaged over a distinct set of frequencies, level matched, and weighted using a frequency-dependent ratio. The two curves are then combined to produce a third curve. The results indicate system performance in a listening space that matches human sensory response and provides means to optimize the sound system for the listening space.

Abstract: Provided are an apparatus and method for controlling an acoustic radiation pattern output through an array of speakers. The apparatus stores a plurality of filter values for forming a plurality of corresponding control patterns set in advance from an input signal. The apparatus forms a compensation control pattern such that signals output through the array of speakers have an intended pattern in consideration of a filter value of a focusing filter. The apparatus selects at least one of the control patterns set in advance to form the compensation control pattern, and processes the input signal using a filter value corresponding to the selected control pattern.

Abstract: A Wide-band Equalization System (“WBES”) based on near- and far-field measurement data. The WBES includes a subwoofer equalizer having an FIR filter together with decimator and interpolator filters for processing low frequency signals. The WBES may also include satellite channels for processing mid- and high-frequency signals, where each satellite channel includes cascaded IIR filters that process mid-frequency and high-frequency signals, respectively. The WBES may also include a DSP that performs the functions required by the IIR and FIR filters.

Abstract: A device and method for improving the performance of an electro-acoustic transducer. An acoustic test signal is generated through the electro-acoustic transducer. The acoustic test signal is measured at multiple points on an ambient surface around the electro-acoustic transducer to create measured acoustic data. Based on the measured acoustic data, an acoustic power frequency response of the electro-acoustic transducer is calculated. A correction impulse response for the electro-acoustic transducer is determined based on the acoustic power frequency response. A correction filter applies the correction impulse response to a sound signal input to generate a sound signal output for playback through the electro-acoustic transducer.

Abstract: A surround experience for headphones can be created using various techniques and tools. The described techniques and tools can be applied separately or in combination. For example, a surround experience can be created by splitting input audio channel signals into high-pass and low-pass signals. The low-pass signals can undergo cross-mixing and artificial reverberation. The artificial reverberation can apply different delay profiles to the low-pass audio signals. The high-pass and low-pass audio signals can be combined to generate output audio channel signals. A surround experience can be crated using a system comprising an input module, a crossover module, a channel cross-mixing module, a reverberation module, and an adder module.

Abstract: A sound signal processing apparatus and a sound signal processing method divide an input signal into a low frequency signal output (11) and a high frequency signal output (10), and delay only the high frequency signal output (10), thereby reducing the temporal shift between the high frequency signal and the low frequency signal. Furthermore, correcting the phase of the low frequency signal output (11) in accordance with a change in phase due to the delay of the high frequency signal output (10) allows to prevent variation in frequency characteristics due to interference at the time of addition of the low frequency signal output (11) and the high frequency signal output (10).

Abstract: A stereophonic sound reproduction system for compensating a low frequency signal and a method thereof, wherein a mono component signal for compensating low frequency signals which are attenuated when removing a crosstalk of inputted left and right signals inputted is calculated using an average value between the left and right signals, left and right compensation gains which are inversely proportional to an absolute value of a power difference value between the first left and right signals, an amplitude of the calculated mono component signal is controlled according to the left and right compensation gains, and thereafter the mono component signal with the controlled amplitude is added to the left and right signals when removing the crosstalk, whereby the left and right signals from which the crosstalk is removed and to which the mono component signal is added are outputted through left and right speakers to thus prevent distortion of the low frequency signals of original stereophonic sound with maintaining a

Abstract: Loudspeaker apparatus comprise commonly some sort of casings or assemblies, in which one or more single speakers are integrated and which are used to convert electrical signals into sound. A loudspeaker apparatus 1 for radiating sound in a hemisphere is disclosed having a center axis 9, the loudspeaker apparatus 1 comprising: a set of midrange drivers 6a, b, c, d, e, f, whereby the midrange drivers 6a, b, c, d, e, f are controlled and/or arranged in a Bessel configuration and are operable to provide a first acoustic field in the hemisphere, a set of tweeter drivers 7a, b, c, d adapted to provide a second acoustic field in the hemisphere, whereby the first and the second acoustic field are respectively arranged symmetrically, whereby the main sound emitting directions of the single tweeter drivers 7a, b, c, d are angled to the center axis 9.

Abstract: A combined multirate-based Finite Impulse Response (FIR) filter equalization technique combines a low-order FIR equalization filter operating at a lower rate for equalization of a loudspeaker-room response at low frequencies, and a complementary low-order minimum-phase FIR equalization filter operating at a higher rate for equalization of the loudspeaker-room response at higher frequencies. The design of two complementary band filters for separately performing low and high frequency equalization, keeps the system delay at a minimum while maintaining excellent equalization performance. Splicing between the two equalization filters, for maintaining a flat magnitude response in the transition region of the two complementary filters, is done automatically through level adjustment of one equalization filter relative to the other. The present invention achieves excellent equalization at low filter orders and hence reduced computational complexity.

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: A canalphone system may include a canalphone housing, and a first high frequency driver carried within the canalphone housing. The system may also include a second high frequency driver carried within the canalphone housing where the second high frequency driver is tuned with the first high frequency driver to deliver lower distortion than a standard canalphone high frequency driver and/or lower distortion than two standard canalphone high frequency drivers that are not tuned with each other.

Abstract: A system for and method of enhancing the low frequency response of a loudspeaker for relatively low input level audio signals and protect the loudspeaker for relatively high input level audio signals is disclosed. The system comprises: a crossover network configured so separate an audio input signal into at least two frequency bands including a low frequency band; and a signal compressor responsive to the energy level of the low frequency portion of an input audio signal in the low frequency band and configured to provide amplification gain on the low frequency portion of the input signal when the energy level of the low frequency portion of the input signal is relatively low so as to enhance the low frequency response of the loudspeaker, and attenuation of the low frequency portion of the input signal when the energy level of the low frequency portion of the input signal is relatively high so as to protect the loudspeaker from being overdriven.

Abstract: Speaker system is constructed of a plurality of speakers each including a bass-reflex type cabinet. The speaker system is designed to differentiate a low-band resonance frequency between the speakers by differentiating inner cubic capacities of the speaker cabinets from each other. Audio signal of a different channel is input to each of the speakers, and only low-frequency signals of all of the channels are added together so that the added result is supplied to all of the speakers. Thus, using the differentiated low-band resonance frequencies, the speaker system permits reproduction of low-pitched sounds with a flat characteristic.

Abstract: Provided are a method and apparatus for controlling a sound field through an array speaker. The method includes calculating a coefficient of a filter that controls sound pressure of an input signal, based on a sound pressure ratio of a suppression area that suppresses sound emitted from an array speaker and an emphasis area that emphasizes the sound, and sound pressure efficiency in the emphasis area, generating a plurality of output signals that focuses the sound to the emphasis area by filtering the input signal based on the calculated coefficient of the filter, and outputting a sound field controlled sound based on the generated plurality of output signals. Accordingly, a listener in a predetermined direction and distance from the array speaker can clearly hear sound, without wearing an earphone or a headset so as to focus the sound only to the listener.

Abstract: The invention maintains optimum sound quality even when a listener changes his/her angle formed between a sound wave emitting axis of an ultrasonic transducer and a front direction axis of the listener with respect to the sound wave emitting axis. An ultrasonic speaker system of the invention includes an angle measuring unit that measures a listener angle as an angle formed between a sound wave emitting axis of the ultrasonic transducer and an axis indicating a listener's front direction; a control information transmitting unit that transmits control information including listener angle information obtained by the angle measuring unit; a control information receiving unit that receives the control information; and a control unit that controls the signal processing performed by a signal processing unit based on the listener angle information included in the control information.

Abstract: A crosstalk cancellation system for preserving quality of sound and the parameter determining method thereof. The system can cancel a two-channel audio signal without compensating a frequency response of an ipsilateral HRTF. The method includes providing an audio signal input device for inputting the two-channel audio signal, modeling a relation between parameters of the crosstalk cancellation system and a head related transfer function (HRTF) to thereby obtain an output transfer impulse response, setting contralateral parts of the output transfer impulse response approximate to zero, setting ipsilateral parts of the output transfer impulse response equal to a delay of ipsilateral head related impulse responses (HRIRs) corresponding to the HRTF, and performing an inverse operation to compute the parameters of the crosstalk cancellation system.

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 speaker that includes a pair of spaced-apart, in-plane mounted drivers connected in series to a network for applying the appropriate frequency component of the electrical input drive signal to each of the drivers is disclosed. A frequency dependent shunting network, such as a low pass filter, is applied to one of the drivers so as to gradually mute the one driver as a desired frequency is approached. The result is an aesthetically pleasing speaker that has dual in-plane drivers and that produces superior sound quality throughout the entire frequency range of the speaker, including those ranges in which dual, in plane, drivers tend to acoustically interfere with each other.

Abstract: A method is provided for computing frequency responses of crossover filters for multi-way loudspeakers. The method prescribes driver coordinates for drivers in the multi-way loudspeaker, prescribes an attenuation function for the sound pressure level at a desired angle, computes the crossover frequencies using a point source model and computes the frequency responses in intervals defined by the crossover frequencies.

Abstract: A woofer-less and box-less loudspeaker system including a plurality of tweeter drivers is provided. The speaker system includes a plurality of drivers, each driver including a front face and a rear face with an axis of symmetry, each driver configured for propagating sound energy along the axis of each driver from the front face, wherein the sound energy includes low frequency and high frequency components; and a support structure for arranging the plurality of drivers in such a way that the axis extending from the rear face of each of the drivers converge in a single point in space, wherein as the sound is propagated along the axis of each driver from the front face, the high frequency components from each driver are evenly spaced and the low frequency components from each driver are reinforced by the low frequency components of adjacent drivers.

Abstract: An adaptor is connectable between a signal processor and a transmission module. The signal processor has three audio channels for feeding respectively a first audio signal, a second audio signal, and a third audio signal having a frequency range lower than a frequency range of the first and second audio signals. The transmission module has two transmission channels capable of transmitting two audio signals to a codeless speaker. The adaptor has three input terminals for receiving the first, second and third audio signals from the signal processor. The adaptor filters at least one of the first and second audio signals for cutting therefrom a lower frequency range comparative to the lower frequency range of the third audio signal, and adds the third audio signal to the one of the first and second audio signals from which the lower frequency range is cut.

Abstract: The quality of music output from audio systems is improved by simulating the effect of low frequency signals in the human ear. This thus allows listeners to perceive the lower frequency signals, even though the speakers may be incapable of providing such low frequency outputs. A method is provided for processing enhancing bass effect in audio signals. The method also results in the bass enhancement being computationally less intensive. The bass effect enhancement techniques are based on the response of sine and cosine transfer functions and on the directional independence of low frequency components. The human ear is unable to resolve directions from low frequency components. The bass effect enhancement technique alternatively is based on response of an exponential transfer function.

Abstract: A multiband DRC system and a method for controlling the same are disclosed. In accordance with the present invention, the system and the method are capable of increasing an overall loudness of an output signal by controlling thresholds and gains of plurality of DRCs included in the multiband DRC system according to frequency bands.

Abstract: An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform: filtering an audio signal into at least three frequency band signals; generating for each frequency band signal a plurality of sub-band signals; processing at least one sub-band signal from at least one frequency band; and combining the processed sub-band signals to form a combined processed audio signal.

Abstract: Apparatuses, methods, and systems directed to enhancing sound quality in audio signal transmissions. Some embodiments of the present invention comprise an audio signal processor operable to boost one or more frequency components of an input signal. In one embodiment, the audio signal processor boosts three frequency components comprising frequency bands centered near 300 Hz, 1.7 kHz, and 5.4 kHz. Other embodiments of the present invention can be used to enhance stereo audio signals comprising a left and a right input signal. The left and the right input signals are filtered and one or more frequency components of the left and the right input signals are boosted by one or more predetermined values. Some embodiments of the present invention comprise one or more integrated circuits comprising one or more analog or digital filters operable to boost one or more frequency components of an audio input signal.

Abstract: A system and method provide at least a single stage optimization process which maximizes the flatness of the net subwoofer and satellite speaker response in and around a cross-over region. A first stage determines an optimal cross-over frequency by minimizing an objective function in a region around the cross-over frequency. Such objective function measures the variation of the magnitude response in the cross-over region. An optional second stage applies all-pass filtering to reduce incoherent addition of signals from different speakers in the cross-over region. The all-pass filters are preferably included in signal processing for the satellite speakers, and provide a frequency dependent phase adjustment to reduce incoherency between the center and left and right speakers and the subwoofer. The all-pass filters are derived using a recursive adaptive algorithm.

Abstract: An automated audio tuning system may optimize an audio system for power efficiency when performing automated tuning of the audio system to optimize acoustic performance. The system may establish any number of different power efficiency weighting factors to provide a balance between acoustic performance and power efficiency during operation. The power efficiency weighting factors may range from representing optimizing power efficiency with constrained optimization of acoustic performance to optimized acoustic performance with minimized regard for power efficiency. For each of the efficiency weighting factors, the system may generate operational parameters, such as filter parameters, to achieve a target acoustic response while maintaining a determined level of power efficiency.

Abstract: A system and method provide at least a single stage optimization process which maximizes the flatness of the net subwoofer and satellite speaker response in and around a cross-over region. A first stage determines an optimal cross-over frequency by minimizing an objective function in a region around the cross-over frequency. Such objective function measures the variation of the magnitude response in the cross-over region. An optional second stage applies all-pass filtering to reduce incoherent addition of signals from different speakers in the cross-over region. The all-pass filters are preferably included in signal processing for the satellite speakers, and provide a frequency dependent phase adjustment to reduce incoherency between the center and left and right speakers and the subwoofer. The all-pass filters are derived using a recursive adaptive algorithm.

Abstract: An audio signal processing apparatus includes: a first filtering unit which outputs an audio signal while attenuating frequency components except for preset frequency components; a detecting unit which detects a sound volume level; an amplitude limiting unit which calculates an amplitude limiting level corresponding to the sound volume level, and which limits a part of a waveform of the audio signal output from the first filtering unit; a second filtering unit which outputs the audio signal output from the amplitude limiting unit while attenuating frequency components except for preset frequency components including a part of the frequency band of the audio signal output from the first filtering unit, and a part of the frequency band of the harmonics; a compressing unit which compresses a dynamic range of the audio signal; and an adding unit which adds the audio signal output from the compressing unit to the input audio signal.

Abstract: A device (30) for adapting an audio input signal (V1n) to a transducer unit (20) comprises: mapping means (10) for mapping input signal components from a first audio frequency range onto a second audio frequency range so as to produce a mapped audio signal (VM), wherein the second audio frequency range is narrower than the first audio frequency range, and wherein the transducer unit (20) has a maximum efficiency at the second audio frequency range, filter means (31) for filtering the input signal (V1n) so as to produce a filtered input signal (V1n?) having a third audio frequency range, and combination means (32) for combining the mapped audio signal (VM) and the filtered input signal (V1n?) so as to produce a transducer signal (VT). The first audio frequency range is preferably contained in the second audio frequency range, while the third audio frequency range may be adjacent the first audio frequency range.

Abstract: A gaming machine for conducting a wagering game includes selective application of surround sound. The selective application of surround sound allows certain key gaming events to be emphasized over other gaming events. The surround sound may be varied to produce certain effects, such as a tactile effect and a gradual building up of suspense and anticipation that increases player excitement and enjoyment. A surround sound decoder may be added in some embodiments to convert monophonic and stereo wagering games into surround sound. The surround sound may be implemented as true surround sound or as virtual surround sound.

Abstract: A band division apparatus obtains an analog low-band signal, an analog intermediate-band signal and an analog high-band signal from a digital sound signal. In the band division apparatus, a digital filter separates the digital sound signal into a digital intermediate-band signal having the intermediate frequency band while the high frequency band and the low frequency band are attenuated, and a digital low-high-band signal having a frequency band combining the low frequency band and the high frequency band while the intermediate frequency band are attenuated. A DA converter converts the digital low-high-band signal into an analog low-high-band signal. Another DA converter converts the digital intermediate-band signal into the analog intermediate-band signal. An analog filter separates the analog low-high-band signal into the analog low-band signal and the analog high-band signal.

Abstract: A signal processing device for supplying sound signals to a first speaker for emitting a sound on a low-pitched sound side and a second speaker for emitting a sound on a high-pitched sound side higher than the low-pitched sound side, includes a control portion that controls a characteristic of the sound signal supplied to the first speaker so that an output level of the first speaker is maintained at a crossover frequency between the first speaker and the second speaker, when an output level of the sound signal supplied to the first speaker is changed.

Abstract: The present invention comprises a circuit having at least not less than three cascade connected phase shift circuits and wherein the phase characteristic of the circuit is determined in such a manner that the phase of the audio input signals is gradually delayed depending on logarithmic increase of frequencies in a range of audible frequencies and a phase delay in a range of 450 degree˜600 degree in total in the range of audible frequencies is caused.

Abstract: Integrated processing of audio/video signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated system includes a main player that captures and processes signals digitally, a dummy display, and a dummy speaker. The dummy display may only have a display panel and a panel driver. The dummy speaker may only have a driving unit and no crossover logic. The main player may have a PC architecture and process all signals digitally for outputting signals tailored for the display device and the individual driving units of the dummy speaker. The integrated system may also provide dynamic signal adjustments based on the surrounding environment. The main player may include a storage device and can process media content stored therein to produce supplemental information to provide an optimal audiovisual experience. This supplemental information can be shared among users over a network connection.

Abstract: A Wide-band Equalization System (“WBES”) based on near- and far-field measurement data. The WBES includes a subwoofer equalizer having an FIR filter together with decimator and interpolator filters for processing low frequency signals. The WBES may also include satellite channels for processing mid- and high-frequency signals, where each satellite channel includes cascaded IIR filters that process mid-frequency and high-frequency signals, respectively. The WBES may also include a DSP that performs the functions required by the IIR and FIR filters.

Abstract: A loudspeaker system includes at least two stand-alone loudspeaker devices (1L, 1R) each comprising a frame (3) provided with a vertically oriented array (5) of first loudspeakers (7) for reproducing sound in a higher frequency range. The system further includes a second loudspeaker (13) for reproducing sound in a lower frequency range, wherein the frequency ranges have a crossover frequency fc and 750 Hz<fc<3000 Hz. The first loudspeakers each have a first radiation surface (9) in each array a central area (9a) of each first radiation surface being situated at a distance d from the central area of neighboring first radiation surface, wherein ?/fe<d<1 m, wherein ?/fe is the wavelength of the reproduced sound at the crossover frequency. The system is able of reproduce sound of high fidelity qualification.

Abstract: A simplified digital implementation of a fourth order Linkwitz-Riley crossover network is provided using approximations and transformations of the classical form. The approximation is particularly beneficial when the crossover frequency is low relative to the digital sampling rate, such as when an audio stream is split between bass and treble at about 30-300 Hz and the sampling frequency is about 100 times the cutoff frequency or higher. Rather than merely cascading two sets of second order filters, such as Butterworth filters, a fourth order transfer function is more directly implemented. Conventional transfer functions are simplified through approximations resulting in the elimination of all except one parameter, c, which is a linear function of the cutoff frequency. Additionally, multipliers are moved in line with the integrator elements.

Abstract: A method and system for surround sound beam-forming using vertically displaced drivers provides a low cost alternative to present external surround array systems. A pair of vertically displaced speaker drivers is supplied with surround and main channel information in a controlled phase relationship with respect to each driver such that the surround channel information is propagated in a directivity pattern substantially differing from that of the main channel information. The main channel information is generally directed at a listening area, while the surround channel information is directed away from the listening area and is substantially attenuated in the direction of the listening area, so that the surround channel information is heard as a diffuse reflected field. An electronic network provides for control of the surround channel phase relationship and combining of main and surround signals for providing inputs to individual power amplifiers for each driver.

Abstract: Loudspeaker apparatus comprise commonly some sort of casings or assemblies, in which one or more single speakers are integrated and which are used to convert electrical signals into sound. A loudspeaker apparatus 1 for radiating sound in a hemisphere is disclosed having a centre axis 9, the loudspeaker apparatus 1 comprising: a set of midrange drivers 6 a, b, c, d, e, f, whereby the midrange drivers 6 a, b, c, d, e, f are controlled and/or arranged in a Bessel configuration and are operable to provide a first acoustic field in the hemisphere, a set of tweeter drivers 7 a, b, c, d adapted to provide a second acoustic field in the hemisphere, whereby the first and the second acoustic field are respectively arranged symmetrically around the centre axis 9 of the hemisphere, whereby the main sound emitting directions of the single tweeter drivers 7 a, b, c, d are angled to the centre axis 9.

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: 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.