Abstract: A dereverberation device includes a reverberation estimation unit for estimating a later reflection component by using information on an impulse response from a signal source to an observation point, a noise estimation unit, and a mixing unit. As a result, it is possible to obtain a high-quality dereverberated signal with a small calculation amount even in a noisy environment.

Abstract: Preserving an audio signal in an audio system includes selecting a first audio signal from a plurality of audio signals. The first audio signal is applied to a first transducer. Mix a portion of the first audio signal with a second audio signal from the plurality of audio signals to provide a mixed audio signal. A gain of the first audio signal that is applied to the first transducer is decreased while a portion of the mixed audio signal is applied to a second transducer to preserve at least a portion of the first audio signal.

Abstract: A processor determines an engagement state of a clutch based on a clutch signal, which is output from a clutch switch connected to a clutch pedal. In a declutching state, the processor lowers weighting gain characteristics that serve as acoustic correcting characteristics used in a sound pressure adjuster. Therefore, even when a change of an engine rotational frequency per unit of time increases during a shift-change operation, the gain of the control signal is prevented from increasing. Accordingly, a gap between real acceleration of the vehicle and a sound effect output from a speaker can be decreased, thereby reducing the possibility that the driver and/or a passenger in the vehicle may feel a sense of strangeness relative to the sound effect.

Abstract: Embodiments of a signal processing system, a method, and fractionally modulated digital delay lines are generally described herein. Other embodiments may be described and claimed. In some embodiments, a fractional address is generated by adding a delay value to a fractional offset value, and input sample values are interpolated based on a fractional portion of the fractional address. A write operation may be performed to the integer portion of the fractional address for each sample period using the interpolated input sample values. Adjusted addresses may be generated when addresses are either skipped of duplicated.

Abstract: A method of compensating audio frequency response characteristics of a portable sound system can be performed using acoustic characteristics of a user measured in real-time. The method of compensating the audio frequency response characteristics can be used by a portable sound system. The method includes generating an acoustic characteristics curve of a user based on a minimum perception level of a user with respect to audible audio frequency bandwidths, generating an acoustic compensation curve of the user based on the acoustic characteristics curve of the user and a predetermined frequency characteristics target curve, and compensating the audio frequency response characteristics of a sound based on the acoustic compensation curve of the user.

Abstract: Auxiliary electromagnetic valve for its incorporation into a main valve of a gas servo valve, which comprises a body with an auxiliary outlet conduit connected to an outlet conduit of the servo valve, a pivoting arm that comprises a closure member of a valve seat of the auxiliary outlet conduit, a ferromagnetic armature that closes the magnetic circuit opening the valve seat, and a return means, supported by a plate, that returns the pivoting arm in order to keep the seating closed. The body is configured by means of a peripheral surface fitted to a cavity of the main valve, a lower surface supported against the cavity by means of fitting means, an intermediate transverse surface on which the valve seat is fitted, a transverse perimetral rim upon which the plate is fixed, and an auxiliary hole that connects the valve seat to the outlet conduit.

Abstract: An active sound effect generating apparatus includes a controller (a fourth acoustic adjuster and a fifth acoustic adjuster) which determines the amplitude of a control signal by adjusting the amplitudes of reference signals (intermediate signals) depending on an engine rotational frequency change [Hz/second] calculated by an engine rotational frequency change calculator and an accelerator opening [%] detected by an accelerator opening sensor.

Abstract: LPF and HPF extract bass and treble ranges, respectively, from an input sound signal, and bass and treble boost circuits perform dynamic range expansion/contraction on the extracted bass- and treble-range sound signals in accordance with input levels of the sound signals. The input sound signal and the sound signals output from the boost circuits are added together. There may also be provided coefficient calculation sections for calculating filter coefficients on the basis of the levels of the sound signals extracted by the LPF and HPF. In this case, the bass and treble boost sections perform, in accordance with the filter coefficients calculated by the corresponding coefficient calculation sections, filter processes for increasing/decreasing the levels of the bass and treble ranges, respectively.

Abstract: Circuit and associated methods for dynamically adjusting the spectral content of an audio signal, which increases the harmonic content through the systematic introduction of amplitude asymmetry. In one embodiment, the method comprises a spectral modification of an analog audio signal in which the high-frequency content is reduced as a function of the signal amplitude and spectral distribution. The audio signal is subjected to a complementary pre-emphasis and de-emphasis of the high frequencies.

Abstract: Bandwidth expansion for audio signals by frequency band translations plus adaptive gains to create higher frequencies; use of a common channel for both stereo channels limits computational complexity. Adaptive cut-off frequency determination by power spectrum curve analysis, and bass expansion by both fundamental frequency illusion and equalization.

Abstract: A sound effect generation device for generating a sound effect corresponding to a physical interaction between objects in a virtual space in which a plurality of virtual objects coexists, includes interaction detection means for detecting an object on which the physical interaction occurs in the virtual space, mechanical quantity calculation means for calculating mechanical quantity acting on the object in accordance with the interaction using a mechanical model for determining, based on physics, behavior between the objects on which the physical interaction acts, and sound effect generation means for generating the sound effect caused in the object by the interaction using the calculated mechanical quantity.

Abstract: A sound field controller includes: a sound field generation section which generates an effect sound signal for giving a sound field effect sound to an audio signal; an acquisition section which acquires a measurement signal indicating sound pressure levels of a direct sound and a reflected sound which are collected when a test sound is emitted in a reproduction environment; an identification section which identifies a maximum reflected sound whose sound pressure level is the maximum in a given time period after a collecting timing of the direct sound from the measurement signal; an adjustment section which adjusts the effect sound signal based on a ratio of the sound pressure level of the direct sound to the sound pressure level of the maximum reflected sound; and an output section which outputs the audio signal input to the input section and the effect sound signal adjusted by the adjustment section.

Abstract: The invention is directed to a display and method for conducting sound system design. The display is configured to simultaneously show a sound system model comprising both audio components and sound environment components, a plurality of parameters and a graphic representation of a calculated sound system performance attribute. In response to receiving input from the user and without any additional user intervention, the display re-displays a newly-calculated sound system performance attribute such as, for example, a graphic representation in the sound system model of a sound path associated with a user-selected reflected sound identifier. A processor also simulates, based on the sound system configuration, the sound performance that a user would expect to hear positioned in an identified listening position of the sound system model.

Abstract: Audio tracks including environmental information are recorded on a DVD, or other playback medium using microphones, vibration sensors, or other devices that capture ambient sound from the original environment in which a performance, presentation or other sound is created. For example, environmental audio tracks can be obtained by placing microphones on the floor, ceiling and walls of an amphitheater during a pop music performance. Any object, item or surface can be a candidate from, or within, which to capture an ambient sound. Sounds can be obtained from sensors mounted to musicians, audience members, theater seats, various rooms within a building, etc. The environmental audio tracks (EATs) can be provided individually or mixed together with one or more other EATs, one or more standard recorded tracks, or otherwise mixed or combined with other sounds. Or the EATs can be processed, effected, etc., as desired.

Abstract: A method for Comfort Noise Generation (CNG) comprising the steps of recording information of Background Noise (BGN); generating white noise samples; and generating Comfort Noise (CN) by applying coefficients that are extracted from said information of BGN on White Noise (WN) samples.

Abstract: Audio data are played back by a personal computer a relationship with a playback process recognized easily. Displays showing the contents of play lists are arranged and displayed so that the contents become continuous in sequence in a playback sequence from content that is currently being played back from a display of an operation element that causes an image of a playback process to appear or from a display showing the content which is currently being played back. When the playback of one piece of content of one play list from among a plurality of play lists is terminated and one piece of content of an other play list is played back, at least the content that is currently being played back and content to be subsequently played back are displayed so that the contents can be distinguished.

Abstract: Each of N audio signals are filtered with a unique decorrelating filter (38) characteristic, the characteristic being a causal linear time-invariant characteristic in the time domain or the equivalent thereof in the frequency domain, and, for each decorrelating filter characteristic, combining (40, 44, 46), in a time and frequency varying manner, its input (Zi) and output (Z-i) signals to provide a set of N processed signals (X i). The set of decorrelation filter characteristics are designed so that all of the input and output signals are approximately mutually decorrelated. The set of N audio signals may be synthesized from M audio signals by upmixing (36), where M is one or more and N is greater than M.

Abstract: The invention provides a method and system for forming an output impulse response function. The method includes the steps of creating an initial impulse response, and dividing the impulse response into a head portion and a tail portion. The tail portion is high pass filtered, and low frequency components of the head portion are boosted. The low frequency boosted and high pass filtered respective head and tail portions are then combined into a modified output impulse response, which can then be used to spatialize an audio signal by convolving it.

Abstract: A sound reproduction apparatus and method enhances a low frequency component in a reproduced sound signal using a reflective sound. The sound reproduction apparatus delays and controls gains of input signals, generates a predetermined number of reflective sound signals to enhance a low frequency component, and outputs a sum of reflective sound signals with the enhanced low frequency component.

Abstract: A method and apparatus for conditioning an audio input signal to enhance perception and reproduction of bass frequencies. Harmonics are generated and combined with a phase-shifted version of the audio input signal. Use of a controlled phase shift reduces or eliminates unwanted introduction of waveform asymmetry or D.C. offset.

Abstract: An impulse response synthesis method is carried out by a dispersion calculation process for calculating a dispersion of phase characteristics in association with a given room based on a volume of the room, a degree of sound absorption of the room, and a distance between a sound source and a receiving point arranged in the room, a noise creation process for creating a noise having the dispersion of the phase characteristics calculated in the dispersion calculation process, a trend addition process for adding a phase trend to the noise created by the noise creation process in accordance with the distance between the sound source and the receiving point and obtaining a phase characteristic of a minimum-phase component from the noise added with the phase trend, and a synthesis process for synthesizing an impulse response based on the phase characteristic of the minimum-phase component, the impulse response being used to create reverberation for the room.

Abstract: Special novelty effects apparatus are provided for inclusion on bicycles providing a synthesizer circuit device having a microprocessor circuit element and a memory chip element or flash drive element for receiving instructions for novelty sound and lighting effects that simulate atypical sounds and lighting as would be expected to emanate from a bicycle, such as sound of a motorcycle throttling in takeoff with simulated tailpipe emissions though lighting effects, or sounds of a truck or airplane, all for the recreational and entertainment application to a rider of a bicycle.

Abstract: An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.

Abstract: A signal processing system, a signal processing apparatus and method, a recording medium, and a program in which high frequency components of a narrow-range signal having suppressed high frequency components can be restored with high accuracy. A narrow-range audio signal having low frequency components is divided into frames by a frame-unit dividing unit, and is spectral-transformed into a signal on a frequency domain by a DCT transform unit. A learning-produced-data table extracts, at an address corresponding to a class code supplied from a classifying unit, high-frequency-component DCT spectrum data. In a spectrum combining unit, the extracted high-frequency-component DCT spectrum data is combined with low-frequency-component DCT spectrum data output from the DCT transform unit. Subsequently, a broad-range audio signal having combined high frequency components and low frequency components is output.

Abstract: An apparatus for replicating an engine and/or exhaust sound of a predetermined vehicle includes files stored in computer-readable form in a memory library of the engine and/or exhaust sound of the predetermined vehicle. A microcomputer operated controller receives engine data such as the RPM of the engine of the personal vehicle and continually selects which files to obtain and use. The controller adjusts the sound pattern to optimally correspond in a preferred way with the RPM of the engine of the personal vehicle and the controller provides an output signal that is amplified and fed to a speaker for audio playback of a replicated sound of the engine and/or exhaust sound of the predetermined vehicle in the personal vehicle. A speaker disposed outside the vehicle includes a conical or cylindrical shape. A diagnostic capability is also disclosed. Connection to an OBD II connector provides the desired engine data.

Abstract: Modifying a panned source in an audio signal comprising a plurality of channel signals is disclosed. Portions associated with the panned source are identified in at least selected ones of the channel signals. The identified portions are modified based at least in part on a user input.

Abstract: A method for reducing noise disturbance associated with an audio signal received through a microphone is provided. The method initiates with magnifying a noise disturbance of the audio signal relative to a remaining component of the audio signal. Then, a sampling rate of the audio signal is decreased. Next, an even order derivative is applied to the audio signal having the decreased sampling rate to define a detection signal. Then, the noise disturbance of the audio signal is adjusted according to a statistical average of the detection signal. A system capable of canceling disturbances associated with an audio signal, a video game controller, and an integrated circuit for reducing noise disturbances associated with an audio signal are included.

Abstract: Microphone are provided to an intake port of an engine and a wall surface of an engine room on the interior side respectively to collect an engine sound. The engine sound is processed by a signal processing portion and output via a speaker provided to an interior of a vehicle. Filters for simulating the noise insulating characteristic in the interior of the vehicle and filters for processing the engine sound to emphasize the driving conditions are provided to the signal processing portion. Filter characteristics of the filters are decided in response to sensed values of an engine revolution sensor 30, an accelerator opening angle sensor, and a speed sensor. The driving conditions are emphasized by filtering the engine sound based on the filter characteristics.

Abstract: A system for enhancing sound quality comprising a filter that square roots the instantaneous amplitude of frequencies in an input signal for generating artificial harmonics corresponding to said frequencies. The system can comprise an automatic level control that momentarily boosts the amplitude of a higher frequency portion of the input signal to emphasize attack transients occurring within the input signal.

Abstract: A method, an apparatus, and a software product to process a plurality of input audio signals. The apparatus accepts a plurality of input signals and includes a multi-input, multi-output reverberator arranged to generate a set of output signals including delayed reverberation components simulating the reverberations a listener is likely to hear in a listening environment. The apparatus further includes a multi-input, two-output filter accepting the outputs of the reverberator and the plurality of input terminals, providing outputs for the left and right ears, and configured to implement a set of head related transfer functions corresponding to a listening environment and a set of directions of a listener in the listening environment.

Abstract: There are included an LPF (3) and an HPF (4) that are connected in parallel to the output of a pre-emphasis circuit (2). There is also included a gain adjusting circuit (6) that performs a gain adjustment of low-pass filter with respect to the frequency band to be passed through the HPF (4). The low frequency components of the frequency band of baseband signals outputted from the pre-emphasis circuit (2) pass through the LPF (3), while the high frequency components pass through the HPF (4). As to the outputs from the HPF (4), the gain of especially the higher part of the frequency band components to be passed through the HPF (4) is suppressed by the gain adjusting circuit (6), whereby the amplitudes of the baseband signals can be limited only for the high frequency range without using a limiter and further the peak values of the baseband signals can be inhibited from exceeding the maximum frequency deviation.

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 system and method is disclosed for reconstructing high frequency components of a digital audio signal using a harmonic enhancer in a baseband integrated circuit of a receiver handset. The original spectrum of the digital audio signal is upsampled in a times two (2) upsample unit to double the size of the bandwidth. A low pass filter then removes a high frequency alias of the original spectrum. The spectrum is then modulated with a first carrier frequency and sent to a first filter bank where a low pass filter and a high pass filter shape the modulated harmonic spectrum. After gain adjustment, the modulated harmonic spectrum is added to a delayed version of the original spectrum. Additional harmonic spectra are similarly created at other carrier frequencies and added to the audio output spectra to reconstruct high frequency components of the audio signal.

Abstract: A sound focusing mechanism for generating a focused source of sound, which can be used to identify the transmission loss of an object or identify areas of acoustic leakage is disclosed. The sound focusing mechanism includes a housing having at least one wall formed from a material having high sound insulating properties. A sound generating device for generating a sound is located within a central cavity within the housing. An actuator assembly is operatively connected to the sound generating device for selectively operating the sound generating device. The wall is configured to limit the transmission of the sound there through such that a focused beam or pulse sound is emitted from the mechanism through the opening.

Abstract: A method, apparatus, and system (48) are disclosed for use in synthesizing an audio performance (50) in which one or more acoustic characteristics, such as acoustic space, microphone modeling and placement, can selectively be varied. In order to reduce processing time, the system utilizes pseudo-convolution processing techniques (54) at a greatly reduced processor load. The system is able to emulate the audio output in different acoustic spaces, separate musical sources (instruments and other sound sources) from musical context; interactively recombine (56) musical source and musical context with relatively accurate acoustical integrity, including surround sound contexts, emulate microphone models and microphone placement, create acoustic effects, such as reverberation (58), emulate instrument body resonance and interactively switch emulated instrument bodies on a given musical instrument.

Abstract: There is provided a sound field adjusting method that can provide an acoustic improvement effect tailored to the characteristics of various acoustic rooms, with small differences in reflection properties between sound receiving points. The diameters of a plurality of columnar reflectors are calculated so as to diffuse sound waves of respective different frequency ranges. An arrangement condition is calculated so that the columnar reflectors having the calculated diameters form a plurality of reflecting surfaces that reflect the sound waves of different frequency ranges in random reflection directions, with random reflection time delays, or in random phases. The plurality of columnar reflectors having respective different diameters are then arranged under the arrangement condition.

Abstract: A method for dynamically adjusting the spectral content of an audio signal, which increases the harmonic content of said audio signal, said method comprising translating an encoded digital signal into data bands, creating a psychoacoustic model to identify sections of said data bands that are deficient in harmonic quality, analyzing the fundamental frequency and amplitude of said harmonically deficient data bands, creating additional higher order harmonics for said harmonically deficient data bands, adding said higher order harmonics back to said encoded digital signal to form a newly enhanced signal, inverse filtering said newly enhanced signal, and converting said inverse filtered signal to an analog waveform for consumption by the listener.

Abstract: A method for digitally processing audio signals to emulate the effects of vacuum tube amplifiers and preamplifiers, musical instrument amplification systems, and distortion effects. By use of an implicit numerical method to estimate the response of a parametrically-controlled non-linear transfer function, non-linear filters, and feedback elements, the dynamic behavior and distortion effects of tube amplification stages are simulated. This provides the capability to reproduce the desired sounds of vintage and modern tube amplifier systems and effects with the conveniences and control associated with digital signal processing systems and software.

Abstract: One embodiment of the present invention is a method for producing a system for describing an electrical network. An output signal of the electrical network is sampled at a frequency that corresponds to the Nyquist criterion for the input signal to the electrical network. A model with a memory is developed for the output signal, which is sampled at a low sampling rate, with this model approximating the output signal. The model is then transformed by suitable interpolation to an interpolated model with a memory. The interpolation results in the model created in this way providing a good approximation to an output signal which is sampled at a high frequency. The resultant system can be used for predistortion.

Abstract: Provided are a musical performance-related information output device and a musical performance system capable of superimposing musical performance-related information on an audio signal without damaging the general versatility of the audio signal. The musical performance-related information output device includes a musical performance-related information acquiring section that is adapted to acquire musical performance-related information related to a musical performance of a performer, a superimposing section that is adapted to superimpose the musical performance-related information on an analog audio signal such that a modulated component of the musical performance-related information is included in a band higher than the frequency component of the analog audio signal generated in accordance with the musical performance manipulation of the performer, and an output section that outputs the analog audio signal on which the superimposing section superimposes the musical performance-related information.

Abstract: A circuit, method, and system for producing artificial ambience effect for an input audio signal, mono, stereo, or surround. The ambience effect enhances artificial reverberation, replaces artificial reverberation, or synthesizes extra audio channels, such as surround channels. The circuit may include a transient reduction module and a reverberation filter. The transient reduction module may be adapted to reduce transients in an input audio signal of one or more channels. The reverberation filter maybe adapted to receive a transient-reduced signal of one or more channels corresponding to the transient-reduced signal.

Abstract: A system and method for optimizing digital audio playback is disclosed. The system and method can prevent playback of a digital audio file in an enhanced mode if the digital audio file does not meet a predetermined criterion, such as a particular bit rate. One example of an enhanced mode is a simulated multi-channel surround sound mode.

Abstract: A method for generating background noise and a noise processing apparatus are provided in order to improve user experience. The method includes: if an obtained signal frame is a noise frame, a high band noise encoding parameter is obtained from the noise frame; weighting and/or smoothing is performed on the high band noise encoding parameter to obtain a second high band noise encoding parameter; and a high band background noise signal is generated according to the second high band noise encoding parameter. A noise processing apparatus is also provided.

Abstract: A gaming system and related methods comprising a gaming device and a voucher mechanism in communication with the gaming device. The voucher mechanism is configured to present various types of vouchers to the player depending on a game outcome. The types of vouchers include, without limitation, a jackpot voucher, a merchandise voucher, a free play voucher, a mystery voucher, a competition entry voucher, and a restricted machine play voucher.

Abstract: In a digital signal processor to perform digital signal process with respect to a music signal and to suppress a signal level to a maximum value when the signal level over the maximum value of processable values is generated by the digital signal processor, a first level correcting device corrects the signal level by multiplying the signal level of the music signal by a correction coefficient so as to make the signal level of the music signal over the maximum value, and a second level correcting device corrects the signal level by multiplying the signal level of the music signal corrected by the first level correcting device by a reciprocal of the correction coefficient.

Abstract: A volume control may be provided with some integrated graphical indication of the presence and/or level of an audio signal that it controls. For example, a slider-type volume control may be provided with a signal meter within the slider itself. A volume control may be provided with a mechanism for showing the non-mute volume control level of a volume when the volume control is muted. For example, a slider-type volume control may display a ghosted slider tab at the level of volume control in effect before muting while at the same time displaying a regular slider tab at a muted level. A user may be able to glance at a volume control and quickly and intuitively understand aspects of the volume control's state such as the level and condition of the signal that is flowing through or blocked by (if muted) the control.

Abstract: Systems and methods provide for the creation of vehicle sound fonts, for utilizing the vehicle sound fonts to augment the engine sounds of a vehicle, and for establishing a network for sharing and controlling the playback of vehicle sound fonts and other media. The systems install in automobiles and play simulated engine sounds of other recorded vehicles in a controlled manner synchronized with the engine spark frequency of the playback vehicle. Automobiles with these systems may network together to share media and to coordinate playback in a synchronized manner among all vehicles of the network.

Abstract: A device designed to be plugged into an electric socket in a vehicle, which socket constitutes part of the electrical system of the vehicle, which device, without any other connection, is able to detect the engine speed of the vehicle and convert this information to sound signals. In at least one embodiment, the device includes at least one device for detecting electrical pulses resulting from the engine speed that are unintentionally emitted from the generator, electric motor or ignition system of the vehicle, in order to record the current engine speed. Further, in at least one embodiment, the device transmits, by way of radio waves, sampled/simulated or synthetically generated noise signals based on the electrical pulses to the radio receiver/sound system of the vehicle in order to play these in or around the vehicle.

Abstract: An information processing apparatus is provided which includes a signal conversion unit for converting an audio signal to a pitch signal indicating a signal intensity of each pitch, a melody probability estimation unit for estimating for each frame a probability of each pitch being a melody note, based on the audio signal, and a melody line determination unit for detecting a maximum likelihood path from among paths of pitches from a start frame to an end frame of the audio signal, and for determining the maximum likelihood path as a melody line, based on the probability of each pitch being a melody note, the probability being estimated for each frame by the melody probability estimation unit.

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.