Abstract: In some embodiments, a virtual bass generation method including steps of: performing harmonic transposition on low frequency components of an input audio signal (typically, bass frequency components expected to be inaudible during playback of the input audio signal using an expected speaker or speaker set) to generate transposed data indicative of harmonics (which are expected to be audible during playback, using the expected speaker(s), of an enhanced version of the input audio which includes the harmonics); generating an enhancement signal in response to the transposed data; and generating an enhanced audio signal by combining (e.g., mixing) the enhancement signal with the input audio signal. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: A wireless device and method may comprise, by way of example, a device and method for receiving wireless transmissions which may include locating data, or authorization data or program data, for determining its location from the locating data, or for determining synchronization for program data, or for ticketing, or for a combination thereof. Authorization data and/or locating data and/or other data may be used to authorize reproduction and/or controlling of received program data, and or for controlling the wireless device. Video program data may be delayed by a number of video frames, preferably an integer number, so as to be substantially synchronized with natural sound. The device and method may determine a location for delaying received program data to be substantially in time alignment with natural sound. A ticketing entity may control a ticket and/or and authorization, and/or may control a remote device thereby.

Abstract: A pre-coding method for improving diversity gain in an Orthogonal Frequency Division Multiplexing (OFDM) system and a transmission apparatus and method are disclosed. The transmission method includes converting an input transmission signal into parallel signals, pre-coding the parallel signals with a predetermined frequency offset, performing Inverse Discrete Fourier Transform (IDFT) on the pre-coded signals, and up-converting the IDFT signals and outputting the up-converted signals to a wireless network. By performing pre-coding on a transmission signal with a frequency offset in an OFDM system, diversity gain can be improved.

Abstract: A sound setting apparatus includes: a manipulating unit capable of making an input manipulation in upward, downward, left, right, and rotation directions; a band setting unit that sets each of individual frequency bands corresponding to low, midrange, and high frequencies in controlling an equalizer, in response to an input manipulation to the manipulating unit in the left and right directions; a level setting unit that sets an output level of each of the bands set by the band setting unit, in response to an input manipulation in the upward and downward directions; a center frequency setting unit that sets a center frequency of each of the bands set by the band setting unit, in response to an input manipulation in the rotation directions; and a display unit that collectively displays setting images to set the band, the output level, and the center frequency on a screen on the same layer.

Abstract: Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals.

Abstract: An array speaker apparatus SParray includes a first radiation means for driving speaker units so that a first sound S1 of a main channel is radiated to a wall surface W1 on the left or right side of a listening position, and a second radiation means for driving the speaker units so that a second sound S2 the same as the first audio signal is radiated directly to the listening position.

Abstract: For controlling the acoustic reproduction of audio data containing audio elements that are periodically repeated, movement data regarding a movement process is detected. The movement process contains recurring events. Reproduction of the audio data is controlled using the movement data in such a way that at least within a certain period, one out of n audio elements that are periodically repeated is reproduced in synchrony with the moment one of the recurring events occurs (synchronization) or is reproduced temporally offset by a given amount of time from the moment one of the recurring events occurs (offset synchronization). The value n represents a positive integer.

Abstract: An effect adding method, includes: applying different gains to a positive side waveform portion and a negative side waveform portion of an audio signal respectively when absolute values of input levels of the positive side waveform portion and the negative side waveform portion are smaller than a predetermined value; producing a higher range component of the audio signal based on a high range component of the audio signal to which the gain is applied, the higher range component being higher in frequency than the high range component; producing a lower range component of the audio signal based on a low range component of the audio signal to which the gain is applied, the lower range component being lower in the frequency than the low range component; and synthesizing an audio signal having an effect sound by adding the audio signal to which the different gains are applied, the higher range component, and the lower range component with each other.

Abstract: A loudspeaker that can be bi-wired or not according to a user's preference comprises a case in which is provided a plurality of audio drivers supplied by a respective plurality of audio networks, the networks being supplied (in turn) by a respective plurality of input terminal pairs, further comprising at least one switch bridging the terminals of different pairs, and adapted to selectively connect the terminals to each other.

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 an environmental ambient noise level detection unit for detecting an environmental ambient noise level contained in an audio signal inputted through sound collection means for collecting a transmission sound at the time of a voice call, a signal level adjustment unit which has a level adjustment function to adjust an output signal level with respect to an input signal level, and an input/output characteristic change function to change an input/output characteristic when adjusting a level in the level adjustment function by means of a control signal, and in which a received sound signal in the case of the telephone call voice is arranged to be an input signal, and a control signal generation unit for generating the control signal for changing the input/output characteristic of the signal level adjustment unit from the environmental ambient noise level detected by the environmental ambient noise level detection unit.

Abstract: A method for determining coefficients of a family of cascaded second order Infinite Impulse Response (IIR) parametric filters used for equalizing a room response. The method includes determining parameters of each IIR parametric filter from poles or roots of a reasonably high-order Linear Predictive Coding (LPC) model. The LPC model is able to accurately model the low-frequency room response modes providing better equalization of loudspeaker and room acoustics, particularly at the low frequencies. Advantages of the method include fast and efficient computation of the LPC model using a Levinson-Durbin recursion to solve the normal equations that arise from the least squares formulation. Due to possible band interactions between the cascaded IIR parametric filters, the method further includes optimizing the Q value of each filter to better equalize the room response.

Abstract: A method and device for processing signals representing speech or audio via a plurality of filters that approximate behaviors of the basilar membrane of human cochlea. Each of the plurality of filters is formed from a mother filter via the dilation and a shift in time and has the similar impulse response of the basilar membrane to the frequency band for which the filter represents. Any process can be conducted and any feature can be extracted in the domain of the filters' outputs for applications, such as noise reduction, speech synthesis, coding, and speech and speaker recognition. Processed signals can be synthesized back to the time domain via an inverse cochlear transform.

Abstract: Disclosed is an audio device that plays back compressed audio signals that can be sufficiently comfortable to be audible even in a play back environment with improved acoustic quality. In order to correct a weak signal component near a frequency with a high output level of a compressed audio, weak-signal component adding unit (121) extracts a specific frequency region based on input audio signals, generates a weak-signal component composed of harmonics signals composed of a harmonic component and the like, and a noise signal and adds the generated weak-signal component to the input audio signal. In order to correct a high-frequency band output level from which compressed audio signals are omitted, high-frequency band audio adding unit (122) generates audio of a high-frequency band composed of a harmonic signal and a noise signal and adds the generated high-frequency band audio to the input audio signal.

Abstract: A control device and an associated method for driving a multi-function speaker supporting a plurality of predetermined functions including at least an audio function and a non-audio function includes a digital signal mixing block and a digital-to-analog block. The digital signal mixing block is arranged for receiving a plurality of digital input signals corresponding to the predetermined functions, respectively, and generating a digital mixed signal according to the digital input signals. The digital-to-analog block is coupled to the digital signal mixing block, for generating an analog driving signal to the multi-function speaker according to the digital mixed signal.

Abstract: A frequency equalization system is provided that substantially equalizes the room frequency responses generated by at least one loudspeaker within a listening area so that the frequency responses in the listening area are substantially constant and flat within a desired frequency range with minimum signal latency. The frequency equalization system may use multiple microphones to measure the audio signals of one or more subwoofers to achieve an improved bass response that is flat across the relevant frequency range. The system employs an algorithm that is a closed-form, non-iterative, mathematical solution and features short computation time with reduced delays.

Abstract: A mobile terminal including an output unit configured to output sound, an equalizer configured to adjust parameters of the sound output by the output unit, a display unit including a touch screen and configured to display a Graphic User Interface (GUI) including a graphical guide that can be touched and moved to adjust the parameters of the sound output by the output unit, and a controller configured to control the equalizer to adjust the parameters of the sound output by the output unit in accordance with a shape of the graphical guide that is touched and moved.

Abstract: A signal processing system enhances an audio signal. The audio signal is divided into audio sub-band signals. Some audio sub-band signals are excised. Other audio sub-band signals are processed to obtain enhanced audio sub-band signals. At least a portion of the excised audio sub-band signals are reconstructed. The reconstructed audio sub-band signals are synthesized with the enhanced audio sub-band signals to form an enhanced audio signal.

Abstract: A front surround sound reproduction system using a speaker array, a front surround sound reproduction system to perform stereo localization using a beam forming speaker array, and a surround sound reproduction method thereof. A front surround sound reproduction apparatus using a plurality of speakers includes a first signal processing unit to adjust a frequency characteristic of each first and second channel signal and to output the adjusted signals to left and right speakers assigned according to a frequency band, a second signal processing unit to generate a plurality of channel signals by copying the first and second channel signals, to adjust a signal characteristic of each of the plurality of channel signals, and to output the adjusted signals to a speaker array in the center, and a speaker unit having a plurality of speakers to reproduce the signals output from the first signal processing unit and the second signal processing unit.

Abstract: A digital audio signal processor uses a matrix transform to produce a multi-channel synthetic reverberation signal based on variably-delayed versions of an input signal. The matrix transform outputs are mixed with the inputs to the signal delaying mechanisms, and some of the outputs are further processed to create the reverberation signal.

Abstract: A speaker system comprises a first speaker (203) and a second speaker (205). A driving circuit receives an audio signal and comprises a first drive circuit (209) generating a first drive signal for the first speaker (203) in response to a first filtering of the audio signal with a first pass band. A second drive circuit (211) generates a second drive signal for the second speaker (205) in response to a second filtering having a second pass band which comprises a frequency band below the first pass band. A delay (213) delays the second drive signal relative to the first drive signal. The sound from the second speaker is directionally radiated with a directional radiation pattern having a notch towards the listening position (111). The system uses the precedence effect and non-direct low frequency audio radiation to ensure that directional cues are predominantly provided by the first speaker (203) which may be small and positioned remote from the second speaker (205).

Abstract: The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described.

Abstract: Provided is a signal processing apparatus, including a filter unit that filters an audio signal created by decimating a portion of frequency components by an all-pass filter and outputs a filtering result thereof as improvement components to improve sound quality of the audio signal, and an adder that generates an improved sound in which the sound quality of the audio signal is improved by adding the improvement components to the audio signal.

Abstract: Provided is an apparatus of separating a musical sound source, which may re-construct mixed signals into target sound sources and other sound sources directly using sound source information performed using a predetermined musical instrument when the sound source information is present, thereby more effectively separating sound sources included in the mixed signal. The apparatus may include a Nonnegative Matrix Partial Co-Factorization (NMPCF) analysis unit to perform an NMPCF analysis on a mixed signal and a predetermined sound source signal using a sound source separation model, and to obtain a plurality of entity matrices based on the analysis result, and a target instrument signal separating unit to separate, from the mixed signal, a target instrument signal corresponding to the predetermined sound source signal by calculating an inner product between the plurality of entity matrices.

Abstract: A headphone device includes a housing having a leakage hole to reduce pressure between a user's ear and the housing, a speaker positioned within the housing, and an audio processing module. The audio processing module is configured to receive an audio signal from an audio device, determine whether the audio signal includes at least a predetermined level of audio having a frequency in a first range of frequencies, transmit a first leakage control signal to a leakage hole valve when it is determined that the audio includes at least the predetermined level of low frequency audio; and transmit a second leakage control signal to the leakage hole valve when it is determined that the audio does not include at least the predetermined level of low frequency audio. The leakage hole valve is configured to close the leakage hole upon receipt of the first leakage control signal and open the leakage hole upon receipt of the second leakage control signal.

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. Method and systems provided for processing enhancing bass effect in audio signals. Said method and systems result in the bass enhancement being computationally less intensive. The bass effect enhancement techniques described 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: An automatic adjustment of subwoofer volume levels relative to the main audio system volume and more specifically, to an improved algorithm control method for sub varying volume which allows adjustment of only the low frequency tones from the subwoofer relative to the main system volume.

Abstract: According to an exemplary embodiment of the present invention, an acoustic manipulator element is provided. The acoustic manipulator element is arrangable relatively to an acoustic source in a manner that the acoustic manipulator element splits frequency selectively sound waves originating from the acoustic source in a reflected and a through component, wherein at least a portion of the acoustic waves of the through component is attenuated by at most 15 dB for acoustic frequencies having a wavelength between 200 Hz and 16000 Hz compared to the sound waves of the acoustic source.

Abstract: A dual-sided radio (100) for enhancing a user's experience is provided. The radio includes a primary transducer (110) on an audio-side of the radio that projects a primary sound, a secondary transducer (120) on a data-side of the radio that projects a mid-high frequency sound, a processor (160) that equalizes (200) audio to the primary transducer and the secondary transducer, and a communication module (130) that receives and transmits communication signals containing the audio. The secondary transducer supplements the primary sound with a mid-high frequency sound (404) to compensate for mid-high frequency loss of the primary sound due to diffraction (300).

Abstract: An interference signal removing apparatus of a radio frequency (RF) receiver includes a low noise amplification unit which performs low noise amplification, a feedback processing unit which removes a necessary signal in a desired band from a signal output from the low noise amplification unit, and performs feedback of the signal from which the necessary signal is removed, and a signal processing unit which transmits a processed RF signal by synthesizing an input RF signal and the feedback signal to the noise amplification unit.

Abstract: A vehicle engine harmonic enhancement system. A method for operating the system includes providing a first engine harmonic enhancement audio signal and equalizing the first engine harmonic enhancement audio signal separately for each of the plurality of loudspeakers to provide individually equalized loudspeaker engine harmonic enhancement audio signals. A vehicle audio system includes a plurality of loudspeakers, an entertainment audio system, and an engine harmonic enhancement system. A method includes equalizing an entertainment audio signal to provide an equalized entertainment audio signal; providing an engine harmonic enhancement audio signal; equalizing the engine harmonic enhancement audio signal separately from the entertainment audio signal to provide an equalized engine harmonic enhancement audio signal; and combining the equalized entertainment audio signal with the equalized engine harmonic enhancement audio signal.

Abstract: A sound processing apparatus includes a processing unit that is configured to acquire an audio signal, perform a correction processing on the acquired audio signal and output the correction-processed audio signal to a sound emitting unit. The correction processing includes an indirect sound adjusting processing in which a given signal processing is performed on an audio signal so as to adjust an influence of an indirect sound to be heard at a sound receiving point, and a frequency characteristic adjusting processing in which a frequency characteristic of an audio signal is adjusted. In the correction processing, a frequency characteristic for the frequency characteristic adjusting processing is determined based on a frequency characteristic of the indirect sound adjusting processing.

Abstract: A consumer audio system has a signal processor coupled for receiving an audio signal. The audio signal is sampled into a plurality of frames. The sampled audio frames are separated into sub-frames according to the type or frequency content of the sound generating source. A time domain processor generates time domain parameters from the separated sub-frames. A frequency domain processor generates frequency domain parameters from the separated sub-frames. The time domain processor or frequency domain processor can detects onset of a note of the audio signal. A signature database has signature records each having time domain parameters and frequency domain parameters and control parameters. A recognition detector matches the time domain parameters and frequency domain parameters of the separated sub-frames to a signature record of the signature database. The control parameters of the matching signature record control operation of the signal processor.

Abstract: An audio system has a signal processor coupled for receiving an audio signal from a musical instrument or vocals. A time domain processor receives the audio signal and generates time domain parameters of the audio signal. A frequency domain processor receives the audio signal and generates frequency domain parameters of the audio signal. The audio signal is sampled and the time domain processor and frequency domain processor operate on a plurality of frames of the sampled audio signal. The time domain processor detects onset of a note of the sampled audio signal. A signature database has signature records each having time domain parameters and frequency domain parameters and control parameters. A recognition detector matches the time domain parameters and frequency domain parameters of the audio signal to a signature record of the signature database. The control parameters of the matching signature record control operation of the signal processor.

Abstract: A semiconductor IC includes an analog audio input circuit, a selecting unit, and an audio processing circuit, which maybe a car navigation chip. The analog audio input circuit includes an RF module and at least one analog audio input module, respectively for providing a first analog audio input signal and at least one second analog audio input signal. The RF module includes an FM/AM broadcast receiving function. The selecting unit is coupled to the RF module and analog audio input module, and outputs a target analog audio input signal according to the first and second analog audio input signals. The audio processing circuit is coupled to the selecting unit, and performs an audio signal process upon the target analog audio signal to generate an audio output signal.

Abstract: A system and method to modify audio communications in an online environment and, in particular, to modifying audio communications using one or more parameters. The system includes a module configured to detect one or more parameters applicable to an incoming audio communication and a weighing engine configured to weigh the one or more parameters and obtain a weighted volume. The system further includes a module configured to transmit the weighted volume to a user participating in a virtual universe. The method determines whether one or more parameters apply to an incoming communication, weighs the incoming communication based on the one or more parameters to determine a weighted volume of the incoming communication, and transmits the incoming communication to a user at the weighted volume.

Abstract: A method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.

Abstract: A device (500) for processing audio data (102) comprises an envelope detector (101) for detecting an envelope of the audio data (102) and a frequency generator (103) for generating an oscillation signal (104) oscillating at a predetermined frequency, wherein the frequency generator (103) is coupled to the envelope detector (101) for supplying the envelope detector (101) with the oscillation signal (104), and the envelope detector (101) detects the envelope of the audio data (102) by sampling the audio data (102) based on the oscillation signal (104).

Abstract: A method of modeling the frequency-dependent input-voltage-to-excursion transfer function of a loudspeaker, comprises, for a plurality of measurement frequencies, measuring a voltage and current and deriving an impedance at the measurement frequency. A frequency-dependent impedance function is derived. By additionally using the blocked electrical impedance and a force factor for the loudspeaker, a frequency-dependent input-voltage-to-excursion transfer function can be calculated. The invention provides a modeling approach which is not based on a parametric model, but computes the transfer functions for a set of frequencies separately. As a consequence, it does not require prior knowledge regarding the enclosure (e.g. closed or vented box) and can cope with complex designs of the enclosure.

Abstract: Provided are a method and apparatus for audio bass enhancement using stereo speakers. By filtering a baseband signal of an input signal whose frequency is lower than a blocking frequency calculated based on the distance between first and second speakers, delaying the filtered signal for a predetermined time period, combining a signal component of the input signal output from the first speaker and a signal component of the delayed signal output from the first speaker and making the combined signal component correspond to the first speaker, and combining a signal component of the input signal output from the second speaker and a signal component of the delayed signal output from the second speaker and making the combined signal component correspond to the second speaker, deep and rich audio bass can be provided by a simple operation without structural modification of speakers with respect to micro speakers in which audio bass reproduction is not conventionally performed efficiently.

Abstract: An audio signal processing apparatus includes: a dividing section dividing each of audio signals of a plurality of channels into a plurality of frequency bands; a phase difference calculating section calculating a phase difference between the audio signals of the plurality of channels, for each of the plurality of frequency bands divided by the dividing section; a level ratio calculating section calculating a level ratio between the audio signals of the plurality of channels, for each of the plurality of frequency bands divided by the dividing section; and an audio signal processing section performing output gain setting with respect to divided signals obtained by the dividing section, on the basis of the phase difference and the level ratio for each of the plurality of frequency bands calculated by the phase difference calculating section and the level ratio calculating section.

Abstract: An audio signal processing apparatus includes a high-frequency components extraction means for extracting high-frequency components higher than a predetermined cutoff frequency from the input audio signal and supplying them to satellite speakers by way of a predetermined high frequency range amplifier, a low-frequency components extraction means for extracting low-frequency components lower than a predetermined cutoff frequency from the input audio signal, a correlation reducing means for reducing the correlation of the high-frequency components and the low-frequency components of the input audio signal and a delay means for delaying the low-frequency components and supplying them to a subwoofer by way of a predetermined low frequency range amplifier.

Abstract: An apparatus for generating a high frequency audio signal that includes an analyzer for analyzing an input signal to determine a transient information adaptively. Additionally a spectral converter is provided for converting the input signal into an input spectral representation. A spectral processor processes the input spectral representation to generate a processed spectral representation including values for higher frequencies than the input spectral representation. A time converter is configured for converting the processed spectral representation to a time representation, wherein the spectral converter or the time converter are controllable to perform a frequency domain oversampling for the first portion of the input signal having the transient information associated and to not perform the frequency domain oversampling for the second portion of the input signal not having the associated transient information.

Abstract: Electronics for altering the audio frequency response of a micro-speaker without modifying the micro-speaker itself are used to provide a selected frequency response of the micro-speaker. The micro-speaker has a resonant peak region, from which the response declines for both higher and lower frequencies. In one embodiment the electronics includes a first circuit for modifying the frequency response curve up to the resonant peak region, and a second circuit for modifying the frequency response curve for audio frequencies higher than this region. Each filter yields an integer multiple of 6 dB per octave slope. In the described embodiment, for approximately correcting presbyacusis (age related hearing loss), a set of high pass filters, low pass filters, and/or high order filters are connected to the micro-speaker to progressively attenuate the frequency response curve as the frequency decreases from 10000 Hz to 100 Hz.

Abstract: A movement control device is provided including audio means for emitting sound audible to a human. At least part of the audible sound includes a repetitive intermittent audible sound provided at such a frequency and/or rate sufficient to allow a person hearing the sound to fall into step with the sound to affect the speed at which the person moves from a first locality to at least a second locality.

Abstract: A player apparatus for playing an input signal after band-extending the input signal includes: an extension controller to determine an extension start band for the input signal in accordance with information relating to the input signal; and a band divider to divide the input signal into a plurality of sub-band signals. The frequency band is extended on the basis of a plurality of the sub-band signals on a side lower than the extension start band, among the plurality of sub-band signals into which the input signal is band-divided by the band divider.

Abstract: Sound can be listened to only in a listening area by maximizing a sound energy difference between a listening area and a non-listening area while maximizing sound radiation efficiency of each sound source. Accordingly, realistic sound can be provided to listeners without causing auditory disturbance to third parties, and maximal sound effects can be obtained with only minimal control.

Abstract: A method and apparatus for processing an audio signal to enhance the perceived lower frequency content of the audio signal when played through an audio output device, includes an input configured to receive an audio input signal, a processor configured to filter the audio input signal to produce a high frequency signal and a low frequency signal, generate an enhancement signal by producing higher frequency harmonics from the low frequency signal, including a process of self convolution, and combine the high frequency signal with the enhancement signal to produce an output signal; and an output configured to receive the output signal and produce an audio output.

Abstract: In a user terminal, a reproducing unit reads out from a music file memory unit a music file selected and input by a user and reproduces the music file. In a normal reproduction, an output unit outputs the reproduced sound as an audio. When a notification information reception unit receives information that needs to be noticed such as an e-mail or a phone call, a notification information analysis unit analyzes the setting of the information or notification sound and determines the degree of importance. A frequency band allocation unit selects from allocation patterns of frequency bands stored in an allocation information memory unit a pattern for the music and for the notification sound according to the degree of importance and the frequency band of the notification sound. An audio processing unit extracts and synthesizes the components of the frequency bands allocated to music and the notification sound, and the output unit outputs accordingly.

Abstract: Two or more acoustic transducers share the same acoustic port in a device. The acoustic properties—such as acoustic impedance and frequency response—of the shared acoustic port are matched to each of the two or more acoustic transducers. To accomplish acoustic impedance matching, a separate back volume is provided for each of the acoustic transducers, matched to that transducer. Frequency response matching can be accomplished by the design of the transducer itself, but also by providing an adjacent element in the acoustic system of the transducer. One transducer may serve as an element in the acoustic system of another transducer. Frequency response adjustment of an individual element may also affect acoustic impedance of the entire port-transducer system.