Abstract: The invention relates to a method for synthesizing an engine noise, in particular of an internal combustion engine, wherein the engine noise is generated by at least one electromechanical transducer, in particular an actuator or a loudspeaker, by means of a signal value corresponding to an electrical transducer excitation signal.

Abstract: A sound enhancement apparatus and method which produce low IMD over a broadband frequency region and performs BSE to offer a sound which is natural to the human ears, are provided. The sound enhancement apparatus includes a preprocessor, a BSE signal generator, and a gain controller. The preprocessor divides a source signal into a high-frequency signal and a low-frequency signal and analyzes the low-frequency signal to obtain prediction information regarding a degree of distortion that will be generated by the low-frequency signal. The BSE signal generator generates a higher harmonic signal for the low-frequency signal as a BSE signal to be substituted for the low-frequency signal, wherein the order of the higher harmonic signal is adjusted based on the prediction information regarding the degree of distortion. The gain controller adjusts a synthesis ratio of the low-frequency signal and the BSE signal adaptively depending on the prediction information regarding the degree of distortion.

Abstract: A low-noise pre-amplifier with an active load element is integrated into a microphone. The microphone has an acoustic sensor coupled to the intrinsic pre-amplifier. A controllable current source is coupled to the intrinsic pre-amplifier and supplies a pre-amplifier bias current. A current source controller is coupled to the current source and controls the amplitude of the pre-amplifier bias current to maintain the intrinsic pre-amplifier at a bias point at which the intrinsic pre-amplifier amplifies microphone signals produced by the acoustic sensor. The intrinsic pre-amplifier may be actively regulated at the pre-determined bias point using negative feedback. Alternatively, the intrinsic pre-amplifier may be set to the pre-determined bias point by sweeping the pre-amplifier bias current for the intrinsic pre-amplifier over a range of currents.

Abstract: The invention provides an audio signal processing system for simulating sound engineering effects. The audio signal processing system may simulate, emulate or model sound engineering effects that may be present in a sample audio signal contained in a sound recording. The audio signal processing system may include an input signal, a first filter system, a nonlinear effect simulator and a second filter system. The input signal may include an audio signal and the sample audio signal. The audio signal may be a signal generated with a musical instrument and the sample audio signal may be a previously processed signal for a sound recording. The first filter system may include a chain of filters configured to condition the audio signal. The nonlinear effect simulator may receive the audio signal processed by the first filter system and modify the audio signal nonlinearly.

Abstract: A portable audio device suitable for reproducing MPEG encoded data includes a plurality of inputs, a data storage, a display, an audio output, at least one processor, and a battery. The plurality of inputs includes a forward input, a play control input, and a random input. The data storage stores compressed digitized audio data. The at least one processor is responsive to selection of at least one of the plurality of inputs to convert selected compressed digitized audio data stored in the data storage for reproduction by the audio output and to provide information to the display.

Abstract: An audio effects chain sequencing apparatus alters the sequence of effects units in an effects chain so that different sequences of effects units can be connected. In one example the audio effects chain sequencing apparatus can have an external input port and an external output port, at least two component ports each having an internal input port and an internal output port, where the at least two component ports receive an input from the external input port, transfer that input in a component sequence to the component ports, and transfer the input to the external output port at the end of the component sequence and having a sequence selection module which selects the component sequence in which the at least two component ports transfer the input.

Abstract: A method and device may color or modify the tone or sound quality of audio input signals. A processor such as a DSP, may apply two or more filters to the audio input signal, each filter comprising a set of filter coefficients. The processor may combine finite impulse response (FIR) filters of the two or more filters into a power-saving filter. A speaker or sound emitter may emitting an output audio signal from the filtered audio input signal. The output audio signal has a different tone quality than that of the input audio signal.

Abstract: In an automatic correction process, automatic correction processing portions 60 are connected to a set reference channel and target channels, respectively, so that test signals will be input to the automatic correction processing portions 60, respectively. A rise detection portion 60a detects a rise in a test signal input to a corresponding channel, so that a value counted by a sample counter 61 is latched by a latch 60b at the rising timing of this test signal. In accordance with a difference between the counted value latched to the reference channel and the counted value latched to the target channel, a time difference is calculated. In accordance with the time difference, a delay time set for a channel delay means of the target channel is automatically corrected.

Abstract: An absolute-value circuit outputs the absolute value of a signal SIN? that corresponds to an input signal SIN. A clipping circuit clips the signal SIN? that corresponds to the input signal, to a positive limit value and to a negative limit value. A first multiplier multiplies the signal SIN? that corresponds to the input signal, by a predetermined coefficient. A first adder subtracts the output signal of the first multiplier from the output signal of the clipping circuit. A second adder sums a signal that corresponds to the output signal of the first adder and a signal that corresponds to the output signal of the absolute-value circuit. A third adder sums the input signal SIN and a signal that corresponds to the output signal of the second adder.

Abstract: A system for generating 3D sound with adjustable source positioning includes a first stage and a second stage, which is coupled to the first stage and to a speaker array that includes a plurality of speakers. The first stage is configured to position a plurality of virtual sound sources through a positioner output. The second stage is configured to generate a 3D signal for the speaker array based on the positioner output. The speaker array is configured to generate a 3D sound stage including the virtual sound sources based on the 3D signal. The first stage may be further configured to reposition the virtual sound sources.

Abstract: Techniques for storing and retrieving processing properties for processing an audio signal in an audio processing system, such as, an audio mixing console are provided. The processing properties specify audio effects and/or audio mixing applied to the audio signal. Type information is established which relates to audio content properties of the audio signal. The type information allows classifying audio signals based on their audio content. Based on the type information, processing parameters are stored in a database and/or retrieved from the database.

Abstract: A stereophonic sound output apparatus and an early reflection generation method thereof. The stereophonic sound output apparatus includes an early reflection generator to implement an early reflection when a 5.1 channel audio signal is down-mixed to a 2-channel audio signal to play back a 5.1 channel audio signal through a 2-channel headphone. The early reflection generator generates early reflections in pairs in which there is an appropriate time difference between the left side reflections and the right side reflections by generating an interaural time difference between two input audio signals and filtering. It is possible to copy the characteristics of early reflections in a real listening room. It is also possible to implement an early reflection similar to a real reflection measured in an apparatus for playing back the 5.1 channel audio signal through 2-channel headphone. A natural 5.1 channel effect may also be obtained using little computation.

Abstract: An acceleration sound generation device for a vehicle using at least one of an acceleration operation of the vehicle, an acceleration force generated by the acceleration operation, and an acceleration state obtained by the acceleration force as a control parameter. The acceleration sound generation device includes a sound generation unit and an acceleration sound delay unit. The sound generation unit generates an acceleration sound in a vehicle passenger compartment in response to an acceleration sound signal that is generated based on the control parameter. The acceleration sound delay unit delays output of the acceleration sound by the sound generation unit such that the acceleration sound changes and is delayed with respect to a change in the control parameter. The acceleration sound delay unit sets a delay degree of the acceleration sound at a rising time of the control parameter separately from at a falling time of the control parameter.

Abstract: A system and method for enhancing the audio experience on a consumer electronic device is disclosed. A system for enhancing the audio experience on a consumer electronic device including a parametrically configurable processing block is disclosed. An all-digital audio enhancement system suitable for embedding into a low cost, low power application specific integrated circuit is disclosed. A method for configuring an audio enhancement system on a consumer electronic device is also disclosed.

Abstract: A headphone system, installed in a portable electronic device for outputting sounds, includes an audio wire for transmitting audio signals, a sound output device for receiving and demonstrating the audio signals transmitted by the audio wire, an audio processing unit for receiving audio data and outputting the audio signals corresponding to the audio data via the audio wire according to a control signal, an automatic reel wound around by the audio wire for rolling up the audio wire, a length detection unit for detecting a rotating distance and a rotating direction of the automatic reel, and a control unit for outputting the control signal to the audio processing unit according to a detection result of the length detection unit, to control operations of the audio processing unit.

Abstract: An audio user interface is provided in which items are represented in an audio field by corresponding synthesized sound sources from where sounds related to the items appear to emanate. An audio cursor, in the form of a synthesised sound source from which a distinctive cursor sound emanates, is movable in the audio field under user control. Upon the cursor being moved close to an item-representing sound source, a related audible indication is generated by modifying the sounds emanating from at least one of that item-representing sound source and the cursor. In one embodiment, this audible indication also indicates the current distance between the cursor and item-representing sound source and also the direction of the latter from the cursor.

Abstract: A circuit device for a three-dimensional sound system is disclosed. The device circuit contains a plurality of resistors, capacitors and transistors. The circuit is an analog circuit creating an actual three-dimensional sound system where the listener can perceive sound coming from different spatial directions. The circuit device accomplishes this by sensing the amplitude and phase difference between the sound signals.

Abstract: A controller device is provided whereby players can experience realistic sounds. The controller device is connected to the main body of an information processing apparatus, wherein the controller device comprises a first speaker for emitting a sound signal in a low frequency range, and a second speaker for emitting a sound signal in a frequency range that is higher in comparison with the first speaker.

Abstract: A sound processing device includes a noise suppression unit configured to suppress a noise component included in an input sound signal, an auxiliary noise addition unit configured to add auxiliary noise to the input sound signal, whose noise component has been suppressed by the noise suppression unit, to generate an auxiliary noise-added signal, a distortion calculation unit configured to calculate a degree of distortion of the auxiliary noise-added signal, and a control unit configured to control an addition amount by which the auxiliary noise addition unit adds the auxiliary noise based on the degree of distortion calculated by the distortion calculation unit.

Abstract: A system and method are provided for processing sound signals. The processing may include identifying individual harmonic sounds represented in sound signals, determining sound parameters of harmonic sounds, classifying harmonic sounds according to source, and/or other processing. The processing may include transforming the sound signals (or portions thereof) into a space which expresses a transform coefficient as a function of frequency and chirp rate. This may facilitate leveraging of the fact that the individual harmonics of a single harmonic sound may have a common pitch velocity (which is related to the chirp rate) across all of its harmonics in order to distinguish an the harmonic sound from other sounds (harmonic and/or non-harmonic) and/or noise.

Abstract: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: A signal processing apparatus is provided. The signal processing apparatus comprises: an inputting section for inputting audio signals on a plurality of channels; an acoustic type acquiring section which is adapted to acquire an acoustic type of an audio signal on at least one channel of the audio signals; and a process controlling section which is adapted to control a characteristic of sound-field effect applied to the audio signals based on the acquired acoustic type.

Abstract: A method and system for concurrently monitoring contact center agents. A user interface executes on a supervisor station and displays multiple agent identifiers, including a first agent identifier and a second agent identifier, on a display device. The supervisor station provides to a three-dimensional spatial audio engine (3DSAE) first audio attribute data identifying a location of the first agent identifier in the user interface with respect to a supervisor reference position, the first audio attribute data including first audio direction data and first audio distance data. The supervisor station receives from the 3DSAE first voice signals corresponding to the first agent identifier, wherein the first voice signals include aural characteristics based on the first audio attribute data, and presents the first voice signals to an output port.

Abstract: A mechanical acoustic energy control system to be located in a room in which one or more audio speakers are also located so that the audio output of the speakers can be distributed throughout the room to enable a listener in the room to be surrounded by sound. A master acoustic resonator attached to the front wall of the room is adapted to vibrate and thereby produce acoustic waves corresponding to the acoustic energy generated by the speakers. A low frequency (e.g., bass) acoustic resonator sits on the floor of the room below the master acoustic resonator. The low frequency acoustic resonator is adapted to vibrate to reflect low frequency acoustic waves produced by the speakers and the master acoustic resonator. A satellite acoustic resonator is located on each of the back and opposing side walls of the room. The satellite acoustic resonators are adapted to vibrate in response to the acoustic waves reflected thereto by the speakers, the master acoustic resonator, and the low frequency acoustic resonator.

Abstract: One example embodiment increases perceived signal strength of a sound signal based on persistence of hearing. When the ear perceives a signal it takes a finite length of time to process it. During that time period the ear does not recognize input and is, in effect, “Turned Off”. In accordance with one aspect of the technology, these periods of “unheard” audio input are utilized so that they supplement the information which the ear sends to the brain. An exemplary system includes a switch, configured to route the audio alternatively to two signal paths—one which includes a delay circuit that delays that audio by an amount equal to the ear's persistence of hearing interval. The system also includes a signal combiner configured to combine outputs from the two signal paths, so as to provide the brain with a signal that it will perceive as being twice as loud as the original.

Abstract: A display, includes: a display section; a first output section; a second output section; and a housing containing the display section, the first output section, and the second output section, wherein, the display section has an aspect ratio with a width more than twice a height, the first output section is arranged along an upper side of the display section so as to face forward and is configured to output a plurality of channels of audio, and the second output section is arranged along a lower side of the display section so as to face forward and is configured to output a plurality of channels of audio.

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: A recording and/or reproducing apparatus includes a microphone, a semiconductor memory, an operating section and a controller. An output signal from the microphone is written in the semiconductor memory and the written signals are read out from the semiconductor memory. The operating section performs input processing for writing a digital signal outputted by an analog/digital converter, reading out the digital signal stored in the semiconductor memory and for erasing the digital signal stored in the semiconductor memory. The control section controls the writing of the microphone output signal in the semiconductor memory based on an input from the operating section and the readout of the digital signal stored in the semiconductor memory.

Abstract: It is expected to provide a display system, display control method and computer program for effectively outputting sound with plural display apparatuses in a two dimensional array. A control apparatus 2 is connected to a two dimensional array display apparatus unit made with display apparatuses 1, 1, . . . configuring a multivision system. A controlling unit 20 of the control apparatus 2 obtains position information for the display apparatus unit in the two dimensional array. Based on the position of each display apparatus 1 in the two dimensional array display apparatus unit, the controlling unit 20 decides the presence or absence of output sound and selects the channel for the output sound.

Abstract: An apparatus for merging a first spatial audio stream with a second spatial audio stream to obtain a merged audio stream comprising an estimator for estimating a first wave representation comprising a first wave direction measure and a first wave field measure for the first spatial audio stream, the first spatial audio stream having a first audio representation and a first direction of arrival. The estimator being adapted for estimating a second wave representation comprising a second wave direction measure and a second wave field measure for the second spatial audio stream, the second spatial audio stream having a second audio representation and a second direction of arrival.

Abstract: This disclosure shows methods and apparatus to emulate vacuum tube amplifiers that have output transformers, to deliver “tube sound”, where the output transformer nonlinearity is emulated. Prior arts that emulated vacuum tube amplifiers either did not include output transformer nonlinearity in their emulation, or did not do it correctly. The disclosed methods and apparatus have the advantage of low cost and flexibility, because the output transformers are included in the emulation.

Abstract: A surround sound virtualization apparatus and method. The surround sound virtualization apparatus may include an audio decoder to perform head-related transfer function (HRTF) filtering, and a time delay unit to provide a time delay to a plurality of output signals of the audio decoder.

Abstract: An information processing apparatus includes a sensor portion, a judgment portion, and an output portion. The sensor portion detects three-dimensional coordinates designated by a spatially-apart detection target object. The judgment portion determines an area designated in advance, that includes the three-dimensional coordinates detected by the sensor portion. The output portion outputs, based on a result of the judgment by the judgment portion, audio corresponding to audio information from a position corresponding to at least two-dimensional coordinates out of the three-dimensional coordinates.

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 of reproducing a stereo output signal (having a left field and a right field) represented as digital samples such that said stereo signal emulates the production of said audio signal from a specified audio source location relative to a listening source location (an indicated location). A left channel signal is produced by convolving an audio input signal with a broadband response file, selected from a plurality of stored files derived from empirical testing, dependant upon the indicated location. A right channel signal is also produced in this way. The left and right channel signals are each duplicated for playing through each of a plurality of displaced left field loudspeakers and displaced left field loudspeakers displaced right field loudspeakers respectively. Apparatus for reproducing a stereo output signal. A data storage facility having a stereo output signal.

Abstract: A parameter transformer generates level parameters, indicating an energy relation between a first and a second audio channel of a multi-channel audio signal associated to a multi-channel loudspeaker configuration. The level parameter are generated based on object parameters for a plurality of audio objects associated to a down-mix channel, which is generated using object audio signals associated to the audio objects. The object parameters have an energy parameter indicating an energy of the object audio signal. To derive the coherence and the level parameters, a parameter generator is used, which combines the energy parameter and object rendering parameters, which depend on a desired rendering configuration.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: An device and method of generating environmental reverberation effects for digital audio signals is presented. The device includes a reverberation controller. The reverberation controller pre-processes one or more predetermined characteristics of a first audio signal to produce a pre-processed signal and generates a plurality of delayed outputs from the pre-processed signal, each output having a predetermined delay. The reverberation controller also produces a plurality of reflection outputs from the plurality of delayed outputs and combines the plurality of reflection outputs to produce a second audio signal having a desired reverberation response.

Abstract: The present invention relates generally to an ultrasonic stimulating pet toy and, more specifically, to a mouse-like pet toy comprising a sound chip assembly that emits ultrasonic vocalizations that replicate those studied in mice. The instant abstract is neither intended to define the invention disclosed in this specification nor intended to limit the scope of the invention in any way.

Abstract: An audio input signal is processed to produce a processed audio output signal. An audio input signal is received as an original signal. The audio input signal is dynamically filtered to produce a first stage signal consisting of a selected frequency band of the input signal. Gain applied to the first stage signal is dynamically controlled in response to a control signal to produce a second stage signal. The control signal is derived from the first stage signal. Processing the original signal in combination with the second stage signal to produce a processed audio output signal. Processing the original signal in combination with the second stage signal and the first stage signal to produce a processed audio output signal.

Abstract: Exemplary embodiments of methods and apparatuses to correlate changes in one audio signal to another audio signal are described. A first audio signal is outputted. A second audio signal is received. The second audio signal may be stored in a memory buffer. The first audio signal is correlated to conform to the second audio signal. The first audio signal may be dynamically correlated to match with the second audio signal while the second audio signal is received. At least in some embodiments, a size of a musical time unit of the second audio signal is determined to correlate the first audio signal. At least in some embodiments, the adjusted first audio signal is stored in another memory buffer.

Abstract: An information processing system includes a display, a display sensor that detects a movement or a rotation of the display, a transducer unit as an earphone unit or a headphone unit, a sound processing part that processes an audio signal so as to localize a sound image in a position outside a head of a listener wearing the transducer unit and listening to sound, and an operation controller that computes an output from the display sensor to obtain a moving direction and a moving distance, or a rotation direction and a rotation angle of the display, and controls sound processing performed by the sound processing part in accordance with a result of the computation so that a positional relation between the display and the head of the listener is mapped as a positional relation between an image display surface and the head of the listener in a virtual viewing space.

Abstract: In many audio playback systems, frequencies below a given cut off frequency are suppressed either due to speaker constraints or safety constraints. For example, some speakers are only capable of generating signals above a certain frequency. Prolonged low frequency sound can cause damage to speakers or other components. An audio driver can be equipped with a phantom bass module which by doubling, tripling and/or quadrupling frequencies below a cutoff frequency can simulate the bass experience. The doubling, tripling and quadrupling methods disclosed provide a low complexity formulation of a frequency doubling, tripling and quadrupling. In addition, the frequency doubling, tripling and quadrupling formulations are easily adapted to multi-rate processing, where computational savings can be very high.

Abstract: In one embodiment, an improved-customizability digital sound relaxation system having a sound card receiving port and a collectable sound card are cooperative to play prerecorded natural or other sounds by depressing one of a plurality of sound selector switches and a sound card selector switch. The new sounds of each collectable sound card customize the library of available sounds to individual taste. In another embodiment, an improved-flexibility digital sound relaxation system having at least two (2) prerecorded sounds stored at first and second memory locations of internal or external/internal memory devices may be selected and combined for concurrent and/or individual replay by depressing one of a plurality of sound selector switches and a combine switch, or by depressing one of a plurality of sound selector switches, a sound card selector switch and a combine switch.

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: An engine sound synthesis system is operable to analyze sound. Operation of the system may include providing an input sound signal to be analysed and determining a fundamental frequency of the input signal from the input signal or from at least one guide signal. Furthermore, the frequencies of higher harmonics of the fundamental frequency are determined, thus determining harmonic model parameters. A harmonic signal based on the harmonic model parameters is synthesized and a residual signal is estimated by subtracting the harmonic signal from the input signal. Residual model parameters are estimated based on the residual signal. Furthermore, a corresponding method for synthesizing a sound signal is described.

Abstract: Psychoacoustic bass audio signal enhancement can be accomplished using a monotonic, asymmetric polynomial distortion. A non-linear process applies a monotonic, asymmetric polynomial distortion function that has continuous first and second derivatives to generate even and odd harmonics of missing fundamental frequencies. This polynomial distortion produces the desired psychoacoustic effect with a fairly rapid roll-off so as to avoid unpleasant aliasing. Moreover, the lack of first-order discontinuities prevents clicks or glitches.

Abstract: A computer-implemented method and system are provided for profiling or analyzing audio data based on an audio parameter. The audio data is divided into audio data into segments. Each segment is characterized by the audio parameter in a defined range. Effects are selected according to the segments meeting criteria for the audio parameter of the effects. A media presentation is authored or generated using the selected effects.

Abstract: A method of processing an audio input signal represented as digital samples to produce a stereo output signal (having a left field and a right field) such that said stereo signal emulates the production of said audio signal from a specified audio source location relative to a listening source location. An audio input signal is received. An indication of an audio source location relative to a listening source location (an indicated location) is received. A broadband response file for each of the left field and the right field is selected from a plurality of stored files derived from empirical testing, dependant upon said indicated location. The audio input signal is convolved with each of the selected left field response file and the selected right field response file. Apparatus for processing an audio input signal.

Abstract: A terminal and a method of reproducing a sound source which changes a reproducing attribute of a sound source based on input information and reproduced the changed sound source to provide an effect sound are provided. The method of reproducing a sound source, includes detecting generation of an input; extracting at least one sampling point with respect to the input while the input continues; analyzing input information based on the at least one sampling point; determining a reproducing attribute with respect to a sound source corresponding to the input information based on the input information; and reproducing the sound source corresponding to the input information according to the determined reproducing attribute.