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 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: A signal processor to produce a simulated reverberation effect based on an input signal and conveying the impression of multiple interconnected resonating spaces. A feedback delay network produces a reverberation tail signal, which is delayed by varying amounts in a delay module. A panning module produces a multi-channel signal based on the reverberation tail signal and its echoes.

Abstract: A method and apparatus for introducing a time-varying time delay randomly into the individual reproduction channels of a sound recording, two in the case of binaural presentation. This emulates the temporal aspect of microphone and/or listener motion. The present invention may be applied as a unidirectional process. No preparation of the source material is required. It can be applied to any multichannel audio signal set. It can process analog or digital signals. The process may be used with headphones, loudspeakers, hearing aids or similar assistive hearing devices.

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: A headphone system includes sound processor which calculates properties of the environment from signals from an internal microphone and an external microphone. The impulse response of the environment may be calculated from the signals received from the internal and external microphones as the user speaks.

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 audio signal processing system is configured to separate an audio signal into a dry signal component and one or more reverberant signal components. The dry signal component and the reverberant signal components can be separately modified and then recombined to form a processed audio signal. Alternatively, the dry signal component may be combined with an artificial reverberation component to form the processed audio signal. Modification of the reverberation signal component and generation of the artificial reverberation component may be performed in order to modify the acoustic characteristics of an acoustic space in which the audio signal is driving loudspeakers. The audio signal may be a pre-recorded audio signal or a live audio signal generated inside or outside the acoustic space.

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: Methods and systems for echo modulation are described. In one embodiment, intensities of a plurality of values in multiple windows of an audio signal may be obtained. The windows may be subject to a periodic boundary condition. A plurality of echo values may be calculated for each of the respective windows. The audio signal may be altered in one or more of the windows using a windowing function and echo values. Additional methods and systems are disclosed.

Abstract: A method for processing a media signal, comprising: receiving, by an audio processing apparatus, an audio signal including a first channel signal and a second channel signal; estimating center sound by applying a band-pass filter to the first channel signal and the second channel signal; obtaining a first ambient sound by subtracting the center sound from the first channel signal; obtaining a second ambient sound by subtracting the center sound from the second channel signal; applying at least one of delay and reverberation filter to at least one of the first ambient sound and the second ambient sound to generate a processed ambient sound; and, generating pseudo surround signal using the center sound and the processed ambient sound is provided.

Abstract: A system and method include obtaining an aural impulse signature at a first location in a simulated space. An aural impulse signature at a second location in a simulated space is also obtained. Aural impulse signature morphing is applied via a computer to aural impulse signatures of the first and second locations to provide an intermediate aural impulse signature between the first and second locations.

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: Embodiments of the present invention allow participants in a networked virtual world to converse naturally as in a real-life situation. Each participant can hear “nearby” participants localized according to their positions in the virtual world, desirably with reverberation and environmental audio effects derived from properties of the virtual world.

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 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 and apparatus of generating a sound field effect is provided. The sound field effect generating apparatus may generate a frequency coefficient that is frequency-transformed from a direct signal, may generate a reflection signal from the frequency coefficient, may generate an output signal using the frequency coefficient and the reflection signal, and may perform an inverse-frequency transform of the output signal.

Abstract: An apparatus for determining a converted spatial audio signal, the converted spatial audio signal having an omnidirectional audio component and at least one directional audio component, from an input spatial audio signal, the input spatial audio signal having an input audio representation and an input direction of arrival. The apparatus has an estimator for estimating a wave representation having a wave field measure and a wave direction of arrival measure based on the input audio representation and the input direction of arrival. The apparatus further has a processor for processing the wave field measure and the wave direction of arrival measure to obtain the omnidirectional audio component and the at least one directional component.

Abstract: A system and method for tuning a speech recognition engine to an individual microphone using a database containing acoustical models for a plurality of microphones. Microphone performance characteristics are obtained from a microphone at a speech recognition engine, the database is searched for an acoustical model that matches the characteristics, and the speech recognition engine is then modified based on the matching acoustical model.

Abstract: Embodiments of the present invention allow participants in a networked virtual world to converse naturally as in a real-life situation. Each participant can hear “nearby” participants localized according to their positions in the virtual world, desirably with reverberation and environmental audio effects derived from properties of the virtual world.

Abstract: There is provided an apparatus for enabling karaoke. The apparatus includes a casing for the apparatus; a microphone array incorporated within the casing, the microphone array being concealed from a user by the casing; a controller coupled to the microphone array to at least process audio signals from the user input at the microphone array; a data storage device coupled to the controller, the data storage device being for storing pre-recorded songs used for karaoke, and for storing songs as sung by the user; and an image capturing device coupled to the controller, the image capturing device being for capturing images of the user while the user is singing, with the captured images of the user being stored on the data storage device.

Abstract: In a sound processing apparatus, a likelihood calculation unit calculates an in-region coefficient and an out-of-region coefficient indicating likelihood of generation of each frequency component of a sound signal inside and outside a target localization range, respectively, according to localization of each frequency component. A reverberation analysis unit calculates a reverberation index value according to the ratio of a reverberation component for each frequency component. A coefficient setting unit generates a process coefficient for suppressing or emphasizing a reverberation component generated inside or outside the target localization range, for each frequency component of the sound signal, on the basis of the in-region coefficient, the out-of-region coefficient and the reverberation index value. A signal processing unit applies the process coefficient of each frequency component to each frequency component of the sound signal.

Abstract: A system for audio processing comprising a room reflection emulation system for emulating sound reflections in a room. A room acoustics emulation system for emulating acoustic properties of the room. A head, shoulder and ear emulation system for emulation sound reflections near the head.

Abstract: A method of processing at least one input signal by a set of binaural filters such that the outputs are playable over headphones to provide a sense of listening to sound in a listening room via one or more virtual speakers, with the further property that a monophonic mix down sounds good. Also an apparatus for processing the at least one input signals. Also a method of modifying a pair of binaural filters to achieve the property that a monophonic mix down sounds good, while still providing spatialization when listening through headphones.

Abstract: Electronic devices having digital reverberators are disclosed, together with a method of reproducing sound for a user with the digital reverberator. The digital reverberator uses digital surface absorption filters positioned in the reverberator to simulate absorption of energy as digital audio data samples are reflected from virtual surfaces. The position of the digital surface absorption filters enables known frequency-dependent surface absorption characteristics of real materials to be directly implemented using the filter coefficients of each digital surface absorption filter. This enables virtual acoustic spaces to be designed quickly without the need for the digital reverberator to be manually tuned for each space.

Abstract: A head-related transfer function (HRTF) convolution method arranged, when an audio signal is reproduced acoustically by an electro-acoustic conversion unit disposed in a nearby position of both ears of a listener, to convolute an HRTF into the audio signal, which allows the listener to listen to the audio signal such that a sound image is localized in a perceived virtual sound image localization position, the method including the steps of: measuring, when a sound source is disposed in the virtual sound image localization position, and a sound-collecting unit is disposed in the position of the electro-acoustic conversion unit, a direct-wave direction HRTF regarding the direction of a direct wave, and reflected-wave direction HRTFs regarding the directions of selected one or more reflected waves, from the sound source to the sound-collecting unit, separately beforehand; and convoluting the obtained direct-wave direction HRTF, and the reflected-wave direction HRTFs into the audio signal.

Abstract: A sound field control device has an input part through which an audio signal is input. A storage part stores a first factor obtained by calculating a proportion of energy of direct sound in total energy of sound collected in an adjustment environment within a predetermined time. A sound field generation part generates a sound field effect sound from the audio signal input through the input part, and outputs the sound field effect sound at a volume corresponding to the first factor. A calculation part calculates a second factor which represents a ratio of an energy of a direct sound to an energy of sound which is collected in a reproduction environment and which contains the direct sound. A correction part corrects the volume of the sound field effect sound based on a ratio between the first factor and the second factor.

Abstract: A noise feedback coding (NFC) system and method that utilizes a simple and relatively inexpensive general structural configuration, but achieves improved flexibility with respect to controlling the shape of coding noise. The NFC system and method utilizes an all-zero noise feedback filter that is configured to approximate the response of a pole-zero noise feedback filter.

Abstract: An input indicates a request for rendering of an audio source in a clip of an audio track according to an effect. Further, a cloned audio source is generated. The cloned audio source is a copy of the audio source. A cloned effect is generated. The cloned effect is a copy of the effect. The cloned audio source is processed according to the cloned effect to generate a rendered output file. The rendered output file is played without processing the audio source according to the effect.

Abstract: In a sound processing device, an index value calculation unit calculates a first index value that follows change of a sound signal at a first following degree and a second index value that follows the change of the sound signal at a second following degree which is lower than the first following degree. An adjustment value calculation unit calculates an adjustment value effective to adjust a reverberation component of the sound signal based on difference between the first index value and the second index value. A reverberation adjustment unit applies the adjustment value to the sound signal.

Abstract: In embodiments of the present invention improved capabilities are described for determining a multi-dimensional sound signature for a location within a hypothetical space, comparing the multi-dimensional sound signature to a known multi-dimensional sound signature, and modifying the hypothetical space such that the similarity between the multi-dimensional sound signature for the location within the modified hypothetical space and the known multi-dimensional sound signature is increased.

Abstract: The present invention relates to audio signal processing, and more particularly to methods and apparatuses for emulating and controlling various features of mechanical spring reverberation in a digital audio processing system. According to certain aspects of the invention, such an emulation is performed so as to enhance or alter the characteristics of a digitally stored or processed audio signal in substantially the same manner as a mechanical spring reverberation system. In one example embodiment, the propagation of energy through a mechanical spring is simulated using dispersive waveguides, wherein left-going and right-going waves are separately processed, and the effects of dispersion and attenuation commuted to the waveguide ends. According to additional aspects, many spring reverberators contain spring elements arranged in parallel, with no coupling between springs.

Abstract: An audio signal is processed by transforming the signal into a frequency domain representation having a plurality of frequency subbands. A decorrelated signal is derived from the frequency domain representation using a phase rotation.

Abstract: A signal processing apparatus and method for providing a spatial impression. The signal processing apparatus applies a reverberation effect to a summed signal formed by summing original signals corresponding to locations of a plurality of sound sources, and removes a correlation from the summed signal, thereby generating reverberation signals corresponding to the locations of the plurality of sound sources. Next, the signal processing apparatus applies panning information derived from the original signals, thereby reflecting location information of the original signals.

Abstract: The present invention provides a method of generating, on a data processing system, a multi-channel audio convolution reverb, said method comprising providing a plurality of impulse responses corresponding to a desired room to be simulated; receiving, in input, multi-channel audio sample data; performing, for each respective audio channel, same channel convolution operation on said respective audio channel with a corresponding impulse response; for each audio channel other than said respective audio channel, performing cross-channel convolution operation respectively with a corresponding cross-channel impulse response; performing combination of the results of the respective convolution operations; and outputting the combination (or summation) result as said output audio channel; wherein, in said performing said cross-channel convolution operation, wherein at least one convolution operation is performed corresponding to a shorter length of impulse response than at least one other convolution operation, prefera

Abstract: Disclosed is an audio reverberation apparatus. The audio reverberation apparatus interacts with or may include a mechanical or acoustic reverberation element. An audio signal is pitch transposing upward and applied to the mechanical or acoustic reverberation element. The resulting audio reverberant signal retrieved from the mechanical or acoustic reverberation element is pitch-transposed downward by the same factor as the upward pitch transposing. This results in the mechanical or acoustic reverberation element requiring smaller dimensions in comparison to a mechanical or acoustic reverberation element where a non-pitch-transposed audio signal is applied.

Abstract: In some embodiments, a method applying reverberation to audio from at least one client of a set of clients which share a virtual environment, including by asserting position data and at least one input audio stream to a server, selecting (in the server) a reverberation filter for each input audio stream in response to the position data and generating wet audio by applying to the input audio an early reverberation part of the selected reverberation filter. Typically, a client applies a late reverberation filter to the wet audio using metadata from the server. In other embodiments, a server selects a reverberation filter for application to audio in response to position data, asserts the audio and metadata indicative of the filter, and a client applies the filter to the audio using the metadata. Other aspects are systems, servers, and client devices configured to perform any embodiment of the method.

Abstract: In an impulse response processing device, a waveform dividing part divides a waveform of an original impulse response into a plurality of blocks on a time axis. A time adjusting part performs a time adjustment to reduce a time difference between successive ones of the blocks. A waveform synthesizing part generates a waveform of a new impulse response by adding the blocks on the time axis after the time adjustment by the time adjusting part. Further, a windowing part performs a windowing process to multiply each block by a window function having a window width corresponding to a width of the block, a function value of the window function decreasing from a center to both ends of the block. The waveform synthesizing part generates the new impulse response using the blocks after the windowing process by the windowing part.

Abstract: The present disclosure provides a digital audio signal processing system that comprises a set of delay lines, allpass and lowpass filters to achieve the reverberation effect. The present disclosure further provides a method for generating and controlling digital reverberations for audio signals. The reverberation generated will have an increasing echo density in the time domain and a faster decay of high frequency signals than low frequency signals. The controlling mechanism of reverberation generation is realized through the extraction of the real environment characteristics.

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: An apparatus for generating an ambient signal from an audio signal includes a compressor for lossy compression of a representation of the audio signal so as to obtain a compressed representation of the audio signal describing a compressed audio signal. The apparatus for generating the ambient signal further includes a calculator for calculating a difference between the compressed representation of the audio signal and the representation of the audio signal so as to obtain a discrimination representation. The apparatus further includes a provider for providing the ambient signal using the discrimination representation. An apparatus for deriving a multi-channel audio signal from an audio signal includes an apparatus for generating an ambient signal from an audio signal, an apparatus for providing the audio signal as a front-loudspeaker signal and an apparatus for providing the ambient signal as a back-loudspeaker signal.

Abstract: A method of processing a 2-channel stereo sound signal into a spatial stereo sound includes generating stereo signals of first and second channels from a first input channel signal and a second input channel signal by applying a time difference of times taken for each channel signal to arrive at two ears, and a sound pressure difference formed between the two ears and varying with respect to a frequency band or constant, to each of the first and second input channel signals, generating a center channel of the first and second channel by adding the first and second input channel signals, generating a surround signal of the first channel by subtracting one of the first and second input channel signals, from the other one of the first and second input channel signals, and generating a surround signal of the second channel by making the surround signal of the first channel out of phase, generating a plurality of reflected sounds from the generated surround signals, and applying the time difference and the sound p

Abstract: A spatial sound generation method and apparatus by which reflected sounds of an input sound signal are generated and by using the reflected sounds, a spatial sound is generated. The method includes delaying an input signal by applying a specified number of a plurality of delay values to the input signal to generate the specified number of a plurality of reflected sound signals, multiplying each of the delayed reflected sounds by a respective predetermined gain value, and generating additional reflected sounds from each of the gain-multiplied reflected sounds through a feedback loop comprises a delay value and a gain value that are specific for its corresponding gain-multiplied reflected sound signal.

Abstract: A speaker virtualization system provides virtual surround sound using a pair of physical loudspeakers. A multiple surround audio channels input is processed using a combination of head related transfer functions and shaped reverberation to provide widening and front/back auditory clues without requiring any kind of interaural path cancellation. The system uses a 360 degree power-response head related transfer function to provide perceptual separation of the reverberant and direct paths, along with discrete, different reverberation for left and right rendering channels to provide envelopment. By eliminating interaural path cancellation, the speaker virtualization system also produces a wider virtual surround sound effect, without dependency on head position and facing.

Abstract: Calculation is performed for sound paths 112-1, 114-1 along which sounds emitted from a sound emitting point 104 in an acoustic space 102 are reflected and delivered to a sound receiving point 106. By the calculation, entering angles eR1, eR2 by which the sound paths enter the front side 106a of the sound receiving point 106 are obtained. Calculation is then performed to obtain angles by which respective speakers 52C, 52L, 52R, 52SR, 52SL of a 5.1 surround system are arranged in a listening room, with the front side 106a of the sound receiving point 106 centered thereon. Audio signals on the respective sound paths are distributed among channels for any two speakers. Consequently, sharp localization of sound images is achieved, requiring less calculation in simulating acoustic characteristics of the acoustic space 102 in which the sound emitting point 104 for emitting sounds and the sound receiving point 106 for receiving the sounds are placed.

Abstract: The present invention is based on the finding that a sound-reproduction system, which can generate a spatial sound impression in a reproduction environment, can be efficiently and intuitively controlled by means of a graphic user interface when an impulse response associated with a spatial direction with respect to the reproduction environment is graphically represented, and when the possibility is created for a user to change the impulse response graphically so that, based on the user input of a change, the changed impulse response can be graphically represented and the changed graphical representation can be detected, in order to control the sound-reproduction system. Since it is system-theoretically possible to describe all known linear signal processing operations by impulse responses, it is possible, with the inventive graphic user interface, to provide a sound mixer, through a graphical representation, with an intuitive access to sound effects depending on the direction.

Abstract: To provide an audio reproduction apparatus in which a general user inputs simple and easy settings so that audio beams of respective channels can be set. When an array speaker is installed in a room, the user inputs the shape of the room into the audio reproduction apparatus. Based on the shape of the room, the audio reproduction apparatus determines a beam control pattern indicating which directions audio signals of the channels should be formed respectively. The audio reproduction apparatus reads beam control data including delay times for forming the beams in the directions from a pattern memory, and automatically sets the beam control data in a DSP. Thus, only when the user inputs the shape of the room, the beams are controlled with a beam control pattern suitable to the room so that multi-channel audio can be reproduced optimally.

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: An audio signal processing system is configured to separate an audio signal into a dry signal component and one or more reverberant signal components. The dry signal component and the reverberant signal components can be separately modified and then recombined to form a processed audio signal. Alternatively, the dry signal component may be combined with an artificial reverberation component to form the processed audio signal. Modification of the reverberation signal component and generation of the artificial reverberation component may be performed in order to modify the acoustic characteristics of an acoustic space in which the audio signal is driving loudspeakers. The audio signal may be a pre-recorded audio signal or a live audio signal generated inside or outside the acoustic space.