Abstract: A method of modeling a room impulse response according to an embodiment of the invention includes receiving a sound pressure signal that is obtained by a microphone when an impulse-type sound source is excited and detecting a room impulse response; determining boundaries between a plurality of intervals of the room impulse response such that the room impulse response is divided into the plurality of intervals on a time domain; and modeling the room impulse response for each of the plurality of divided intervals using at least two different modeling schemes.

Abstract: Improved techniques for presenting sound effects at a portable media device are disclosed. The sound effects can be output as audio sounds to an internal speaker, an external speaker, or both. In addition, the audio sounds for the sound effects can be output together with other audio sounds pertaining to media assets (e.g., audio tracks being played). In one embodiment, the sound effects can serve to provide auditory feedback to a user of the portable media device. A user interface can facilitate a user's selection of sound effect usages, types or characteristics.

Abstract: An audio processing circuit includes a crosstalk cancellation circuit that is advantageously simplified for use in audio devices that have closely-spaced speakers. In particular, crosstalk filtering as implemented in the circuit assumes that the external head-related contralateral filters are time-delayed and attenuated versions of the external, head-related ipsilateral filters. With this assumption, the circuit's crosstalk filtering is configurable for varying audio characteristics, according to a small number of settable parameters. These parameters include configurable first and second attenuation parameters for cross-path signal attenuation, and configurable first and second delay parameters for cross-path delay. Optional sound normalization, if included, uses similar simplified parameterization. Further, in one or more embodiments, the audio processing circuit and method include or are associated with a defined table of parameters that are least-squares optimized solutions.

Abstract: Test system and method for analog-to-digital converter (ADC) based on a loopback architecture are provided to test an M-bit ADC. In the invention, an N-bit digital-to-analog converter (DAC) converts a digital input to a basic test signal, a segmentation circuit scales the basic test signal and superposes it with segmentation DC levels for providing corresponding segmented test signals, such that the ADC converts the segmented test signals to reflect result of testing. With the invention, practical loopback architecture of low-cost can be adopted for testing.

Abstract: A reverberator for reverberating an audio signal includes a feedback delay loop processor for delaying at least two different frequency subband signals representing the audio signal by different loop delays to obtain reverberated frequency subband signals.

Abstract: An electrical signal derived from an audio source is passed through an amplifier and is then divided into two discrete signal paths. The signal traveling along the first path travels through a high pass filter and is then split so that a portion is delayed and a portion passes through a harmonics generator so that the harmonics are advanced ahead of the delayed portion from the high pass filter. The split circuit is then combined in a summing amplifier. The second portion of the signal is also delayed before it also travels to the mixer. The combined signal is passed through an amplifier and then fed into an output transducer. The use of a delay both on the input signal and the portion of the signal which passed through the high pass filter output just before the summing amplifier enables the harmonics to not be masked by the lower frequencies, making the effect more apparent.

Abstract: A signal processing device includes a generation unit that generates a second signal from a first signal that is obtained by down mixing two signals; a mixing coefficient determination unit that determines, based on a value L and a value ?, a mixing degree for mixing the first signal and the second signal; and a mixing unit that mixes the first signal and the second signal based on the mixing degree determined by the mixing coefficient determination unit. The generation unit includes a first filter that generates a low frequency band signal in the second signal, from a low frequency band signal in the first signal; and a second filter that generates a high frequency band signal in the second signal, from a high frequency band signal in the first signal. The first filter is a filter unit which, for a complex-number signal, de-correlates an input signal and adds a reverberation component by using a delay unit and an all pass filter, and the processing unit is a filter unit different from the first filter.

Abstract: A method of acoustically rendering a virtual environment is described. The method includes receiving a subset of polygons derived for an acoustic display from a set of polygons generated for a graphical display. Acoustic reflections are determined from a sound source that bounce off of polygons in the subset of polygons to a listener position in the virtual environment. It is determined whether a polygon in the subset of polygons causes an occlusion of the sound source at the listener position, and a play list of sounds is generated based on the reflections and occlusions.

Abstract: An acoustic processing device, includes an inputting section which receives a first audio signal and a second audio signal, a first input signal amplifying section which amplifies the first audio signal at an amplification factor, a second input signal amplifying section which amplifies the second audio signal at an amplification factor, a first initial delayed amplifying section, a first adding section which adds the first audio signal amplified by the first amplifying portion and the first audio signal amplified by the first input signal amplifying section, a second adding section which adds the first audio signal amplified by the second amplifying portion and the second audio signal amplified by the second input signal amplifying section, a first supplying section which supplies an audio signal output from the first adding section to a first sound emitting section, and a second supplying section which supplies an audio signal output from the second adding section to a second sound emitting section differen

Abstract: A system and method for Acoustic Echo Cancellation. The system and method include a subband affine projection filter and a variable step size controller configured to cancel an estimated echo from a near-end signal. The system and method also include a divergence detector adapted to reset the subband affine projection filter in response to determining a divergence is occurring. Additionally, the system and method include a double talk detector adapted to transmit a signal to mask an output signal when double talk is detected.

Abstract: Reverberation parameters for one or more positions of interest are derived from graphics data used for displaying a computer-generated environment. For each position of interest for which reverberation parameters are desired, environmental parameters including distances and the hardness of features in a range of interest and at points on cubemap faces are automatically determined from the graphics data. The environmental parameters are stored with the graphics data and associated with each position of interest. Upon rendering of the computer-generated environment, reverberation property set values usable by a reverberation engine are calculated or interpolated between predetermined values according to the environmental parameters.

Abstract: A multi-channel audio decoder reconstructs multi-channel audio of more than two physical channels from a reduced set of coded channels based on correlation parameters that specify a full power cross-correlation matrix of the physical channels, or merely preserve a partial correlation matrix (such as power of the physical channels, and some subset of cross-correlations between the physical channels, or cross-correlations of the physical channels with coded or virtual channels).

Abstract: A sound system design/simulation system provides a more realistic simulation of an existing venue by matching a measured reverberation characteristic of the existing venue and adjusting one or more acoustic parameters characterizing the model such that a predicted reverberation characteristic substantially matches the measured reverberation characteristic.

Abstract: Various embodiments of the present invention are directed to methods for dereverberation of audio generated in a room. In one aspect, a method for dereverberating reverberant digital signals comprises transforming a reverberant digital signal from the time domain into Fourier domain signals using a computing device, each Fourier domain signal corresponding to a subband. For each subband of the Fourier domain signal, the method computes autoregressive model coefficients of the reverberation with the current and previous magnitudes of the Fourier digital signal, and inverse filters the magnitude of the Fourier domain signal using the computing device, based on the autoregressive model coefficients and previous magnitudes of the Fourier digital signal. The method includes inverse transforming the Fourier domain signals with filtered magnitudes into an approximate dereverberated digital signal.

Abstract: This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

Abstract: An acoustic modeling system and an acoustic modeling method use beam tracing techniques that accelerate computation of significant acoustic reverberation paths in a distributed virtual environment. The acoustic modeling system and method perform a priority-driven beam tracing to construct a beam tree data structure representing “early” reverberation paths between avatar locations by performing a best-first traversal of a cell adjacency graph that represents the virtual environment. To further accelerate reverberation path computations, the acoustic modeling system and method according to one embodiment perform a bi-directional beam tracing algorithm that combines sets of beams traced from pairs of avatar locations to efficiently find viable acoustic reverberation paths.

Abstract: The sound field reproducing apparatus includes: a signal obtaining unit which obtains input audio signals of plural channels; an effect adding unit which gives an effect on an output to a sound field; and an effect amount control unit which controls an amount of the effect based on a characteristic of the input audio signal. The sound field reproducing apparatus receives input audio signals of plural channels from a medium such as a DVD, and gives a sound effect to them to output. The amount of the effect given to the input audio signal is controlled based on the characteristic of the input audio signal, especially the cross correlation of the reverberation component between the channels.

Abstract: An apparatus and method for artificial reverberation having one or more input summing nodes receiving an input signal and which each feed a first series of delay elements and the output of which is thereafter summed, weighted, and fed back to the input summing nodes. Also, a feedback summing node which further feeds a second delay element which feeds an output with a signal which has multi-dimensional or multi-delay reverberation relative to the input signal. An alternative embodiment further feeds one or more of the first delay element outputs weighted via an output gain to an output summing node which also sums the output of the second delay element and provides a summed output having a multi-dimensional or multi-delay reverberation. The embodiment is easily and economically implemented via conventional integrated electronic circuits, a combination of analog and digital electronics, digital signal processors, microprocessors, micro-controllers, or computer algorithms.

Abstract: A user selects a plurality of pieces of music data desired to be reproduced concurrently, at an input unit of an audio processing apparatus, from music data stored in a storage device. A reproducing apparatus reproduces selected music data respectively and generates a plurality of audio signals under the control of a control unit. An audio processing unit performs allocation of frequency band, extraction of a frequency component, time-division, periodic modulation, processing and allocation of a sound image, to respective audio signals under the control of the control unit. Then the audio processing unit attaches segregation information of audio signals and information on the degree of emphasis to respective audio signals. The down mixer mixes a plurality of audio signals and outputs as an audio signal having a predetermined number of channels, then an output unit outputs the signal as sounds.

Abstract: A sound field controlling device for supplying audio signals to a plurality of speakers provided in a space to form a sound field in the space, includes a measuring unit which measures levels of indirect sounds, which are outputted from the speakers, reflected from a wall surface of the space, and reach a listening position respectively, a reverberation applying unit which generates a reverberation simulation signal for reinforcing the indirect sounds on the basis of the audio signals, and a reverberation balance adjusting unit which controls the level of the reverberation simulation signal and supplies the controlled reverberation simulation signal to the corresponding speakers on the basis of the levels of the indirect sounds outputted from the speakers so that respective synthesized levels of the indirect sounds and the reverberation simulation signal are balanced between the speakers.

Abstract: A system and method for providing microphonic isolation on a transmission line. The transmission line has a first part and a second part. The first part of transmission line carries a data signal and a microphonic signal. The microphonic signal has frequencies that include those in a range of substantially 20 Hz to substantially 20 kHz. The system includes an isolation apparatus. The isolation apparatus has an input in electrical communication with a first part of the transmission line, an output in electrical communication with the second part of the transmission line, and a filter in electrical communication with the input and the output. The filter is arranged to substantially remove the microphonic signal received at the input from first part of transmission line and pass the data signal to the output.

Abstract: An audio apparatus includes a decay pattern generator configured to generate a decay pattern to be applied in an early reflection region, and generate an audio signal to which a decay pattern is applied by convoluting the generated decay pattern with PCM raw data of a sound source, a reverberation generator configured to generate reverberation from the audio signal to which the decay pattern is applied, and an adder configured to generate an output signal having a sound enhancement effect by adding the PCM raw data to an output of the reverberation generator.

Abstract: A method operates a hearing device and an associated hearing device has at least one microphone for recording an ambient sound and an earpiece emitting an output sound. A fricative of the ambient sound is reproduced in a temporally lengthened form by the hearing device in the output sound. Speech intelligibility for the hearing-impaired can be improved by the lengthening of fricatives.

Abstract: Provided is a multi-channel acoustic signal processing device by which loads of arithmetic operations are reduced. The multi-channel acoustic signal processing device includes: a decorrelated signal generation unit, and a matrix operation unit and a third arithmetic unit. The decorrelated signal generation unit generates a decorrelated signal w? indicating a sound which includes a sound indicated by an input signal x and reverberation, by performing reverberation processing on the input signal x. The matrix operation unit and the third arithmetic unit generate audio signals of m channels, by performing arithmetic operation on the input signal x and the decorrelated signal w? generated by the decorrelated signal generation unit, using a matrix R3 which indicates distribution of a signal intensity level and distribution of reverberation.

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 method is provided for estimating a reverberation signal component of an acoustic signal detected by a microphone where the acoustic signal is comprised of a direct sound component and a reverberation signal component. A method for dereverberation of an acoustic signal is further provided.

Abstract: A karaoke machine with an integrated computer interface device is disclosed. The karaoke machine includes at least one microphone input having an audio output. A computer interface device configured and arranged to receive digital audio and convert it to analog audio. At least one speaker is also included. A summing node is provided to combine the analog audio from the computer interface device with the audio output of the microphone input together into combined audio and transmit the combined audio to the speaker.

Abstract: Microphones are provided at an air inlet of the engine and a vehicle-cabin-side wall surface of an engine room, and engine sounds are picked up. The engine sound is processed by a signal processing section, and the processed engine sound is output from a speaker provided in a vehicle cabin. The signal processing section is provided with a filter which simulates a sound insulation characteristic of the vehicle cabin and a transformation section for processing the engine sound according to driving condition. A spectrum transformation characteristic of the transformation section is determined according to values detected by a vehicle speed sensor, an engine speed sensor, and an accelerator depression sensor, and a spectrum of the engine sound is transformed by means of specification of the spectrum transformation characteristic, thereby enhancing an engine sound.

Abstract: A calculating unit (201) arranges a three-dimensional spatial body containing a listening point within a virtual space in a voice processor (200). The three-dimensional spatial body is constituted by a group of predetermined unit figures. The calculating unit (201) acquires a distribution of a region and an amount of overlap between a region occupied by the three-dimensional spatial body and a region occupied by predetermined objects. A modifying unit (202) modifies reflected sounds of sounds by the predetermined objects based on the distribution of the region of overlap and the amount of overlap acquired by the calculating unit (201). An outputting unit (203) outputs the reflected sounds modified by the modifying unit (202).

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: A sound system design/simulation system provides a more realistic simulation of an existing venue by matching a measured reverberation characteristic of the existing venue and adjusting one or more acoustic parameters characterizing the model such that a predicted reverberation characteristic substantially matches the measured reverberation characteristic.

Abstract: In a case of transient audio input signals, in a multi-channel audio reconstruction, uncorrelated output signals are generated from an audio input signal in that the audio input signal is mixed with a representation of the audio input signal delayed by a delay time such that, in a first time interval, a first output signal corresponds to the audio input signal, and a second output signal corresponds to the delayed representation of the audio input signal, wherein, in a second time interval, the first output signal corresponds to the delayed representation of the audio input signal, and the second output signal corresponds to the audio input signal.

Abstract: A converter and conversion method are disclosed for converting N channel audio input channels into M channel audio output channels, wherein a processor is used for applying a transfer function to a signal received on an input channel to obtain reverberation components of a calculated output channel, wherein said transfer function is a simplified transfer function matching a selected subset of a set of local maxima of a measured reverberation when applied to a corresponding stimulus.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: A system including at least one system microphone, a processor connected to computer readable memory and the at least one system microphone, a measurement microphone connected to the processor, at least one audio speaker, a signal generator connected to the processor and configured to produce signals from the at least one audio speaker for both the measurement microphone and the at least one system microphone to measure the response of the at least one audio speaker, and a device manager application executable from the computer readable memory and configured to perform the steps of calibrating the at least one audio speaker with the measurement microphone using the signals of the signal generator based on the response of the at least one audio speaker thereby creating at least one calibrated audio speaker and calibrating the at least one system microphone with the at least one calibrated audio speaker using the signals of the signal generator.

Abstract: A stereophonic sound reproduction device has a sound amplifying device that amplifies multi channel surround sound signals for respective channels and outputs the multi channel surround sound signals to a plurality of speakers that are located surrounding a listening position, and a rear reverberation creation device, in the rear reverberation creation device, convolution computations are performed on direct sound or initial reflected sound contained in the sound signals of the respective channels of multi channel surround sound signals, and reverberation groups that have flat frequency characteristics and have non-uniform delay time intervals, and that have different delay time intervals for the respective channels while having the same envelope characteristics for the respective channels are created and output for the respective channels.

Abstract: An impulse response processing apparatus is composed of a waveform divider, a time adjuster, an interpolation processor and a waveform synthesizer. The waveform divider divides an impulse response into a plurality of base blocks on a time axis. The time adjuster increases a time difference between two adjacent ones of the plurality of the base blocks. The interpolation processor generates an interpolation block. The waveform synthesizer generates a new impulse response by arranging the interpolation block between the two adjacent base blocks generated through adjustment of the time adjuster.

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

Abstract: A digital signal processing apparatus in which a first digital filter reproduces that part of an impulse response that responds fast, and a decimation filter converts the output of a delay device of the first digital filter to a digital signal having a sampling rate of ½. The digital signal is supplied to the second digital filter that reproduces that part of the impulse response that responds slowly and outputs data representing the response characteristic of this part of the impulse response. An interpolation filter converts an input signal to a signal having the same sampling rate as the digital audio signal input to the digital signal processing apparatus, and the output signal of the interpolation filter is supplied to an adder circuit.

Abstract: The present invention provides a sound reproducing system and a sound reproducing apparatus that can provide a high realistic sensation to a user, without having to do a troublesome task on the user's side. A surround-sound system (100) includes: an array speaker system (20) that is formed with speaker units SPU having the same characteristics; and a signal processing apparatus (120) that drives the speaker units SPU independently of one another and amplifies an audio signal.

Abstract: A system reduces echoes in an audio system by de-correlating audio signals. Adaptive filters provide compensation based on the de-correlated signals. A controller controls the de-correlation based upon the adaptation state of the adaptive filters.

Abstract: A combination equalizer and calibrator circuit assembly includes a calibrator formed of a frequency control circuit, a signal generator, an amplifier circuit connected with a microphone, a compare circuit and a display circuit and connected to the front end of an equalizer set. Calibration is made through the calibrator prior to audio system reproduction, wherein the frequency control circuit provides a reference frequency and different test frequencies for causing the signal generator to generate respective sound signals for output through a speaker; the microphone picks up these sound signals for comparison with respective reference values by the compare circuit; and the display circuit displays the respective comparison results. By means of adjustment through respective adjust circuits, signals of test frequencies are standardized so that the music played through the audio system is well calibrated without sound spectrum distortion due to space or audio system discrepancy.

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: The present invention provides improvements to prior art audio codecs that generate a stereo-illusion through post-processing of a received mono signal. These improvements are accomplished by extraction of stereo-image describing parameters at the encoder side, which are transmitted and subsequently used for control of a stereo generator at the decoder side. Furthermore, the invention bridges the gap between simple pseudo-stereo methods, and current methods of true stereo-coding, by using a new form of parametric stereo coding. A stereo-balance parameter is introduced, which enables more advanced stereo modes, and in addition forms the basis of a new method of stereo-coding of spectral envelopes, of particular use in systems where guided HFR (High Frequency Reconstruction) is employed. As a special case, the application of this stereo-coding scheme in scalable HFR-based codecs is described.

Abstract: A method and system for applying reverb to an M-channel down-mixed audio input signal indicative of X individual audio channels, where X is greater than M. Typically, the method includes steps of: in response to spatial cue parameters indicative of spatial image of the downmixed input signal, generating Y discrete reverb channel signals, where each of the reverb channel signals at a time, t, is a linear combination of at least a subset of values of the individual audio channels at the time, t, and individually applying reverb to each of at least two of the reverb channel signals, thereby generating Y reverbed channel signals.

Abstract: A multi-channel audio decoder provides a reduced complexity processing to reconstruct multi-channel audio from an encoded bitstream in which the multi-channel audio is represented as a coded subset of the channels along with a complex channel correlation matrix parameterization. The decoder translates the complex channel correlation matrix parameterization to a real transform that satisfies the magnitude of the complex channel correlation matrix. The multi-channel audio is derived from the coded subset of channels via channel extension processing using a real value effect signal and real number scaling.

Abstract: The present invention provides a sound field correcting system and a reverberation adjusting apparatus capable of correcting a reverberation time characteristic by an easy operation without setting a complicated parameter. A surround system 100 has a sound reproducing apparatus 120 for adjusting a reverberation component of sound on the basis of a reverberation characteristic of a listening room 10 and amplifying the sound by a speaker system 130; and a microphone 140 for collecting the amplified sound in a specific listening position in the listening room 10.

Abstract: An electronic reverberation system employs a processor to produce a plurality of delays samples that are added to a direct signal to produce reverberant sound. The disclosed system generates or employs a list of gain value pairs that are produced based on control settings or are provided as fixed coefficients. The processor generates reverberation samples by applying these coefficients to delay samples and summing their amplitudes to produce reverberation waveform samples. The reverberation waveform samples are added to the direct signal.

Abstract: A method and apparatus for extracting and changing the reverberant content of an input signal is provided. The method of estimating an impulse response for a reverberant system includes sampling a signal from a reverberant system, having an impulse response, into a sequence of blocks, for each block of the sequence, determining a magnitude in dependence upon a magnitude of a previous block, forming an estimate of the impulse response using determined magnitudes as parameters for a finite impulse response filter.

Abstract: Disclosed is an audio system including a group of loudspeakers that form a sound field by delivering into a single space sound signals passed through respective ones of a plurality of sound signal channels. This audio system is comprised of two characteristic-variable equalizers that are cascaded to each other to constitute a part of the sound signal channels; a sound field characteristics detecting part for supplying test signals through the sound signal channels and detecting sound pressure in the sound field and thereby obtaining a sound pressure signal; and a characteristics adjusting part for adjusting, based on the sound pressure signal, equalizing characteristics of the characteristic-variable equalizers individually and with respect to each of the sound signal channels. The sound field characteristics detecting part selectively generates test signals of different bands.