Audio CAPTCHA Using Echo

A system for determining that a user is either a human and non-human may comprise an interactive voice component and an audio validation component. The audio validation component may implement a test to determine that the user is one of human and non-human. The test may comprise an echo perturbation effect applied to at least a portion of a challenge phrase to form a modified challenge phrase. The test may further comprise the modified challenge phrase issued to the user, a response received from the user, and an evaluation of the response. When the response is a correct response to the challenge phrase, the user is designated as human, and when the response is an incorrect response to the challenge phrase, the user is designated as non-human. The interactive voice component may be an IVR system, and the audio validation component comprises an auditory CAPTCHA.

Generating Binaural Audio in Response to Multi-Channel Audio Using at Least One Feedback Delay Network

In some embodiments, virtualization methods for generating a binaural signal in response to channels of a multi-channel audio signal, which apply a binaural room impulse response (BRIR) to each channel including by using at least one feedback delay network (FDN) to apply a common late reverberation to a downmix of the channels. In some embodiments, input signal channels are processed in a first processing path to apply to each channel a direct response and early reflection portion of a single-channel BRIR for the channel, and the downmix of the channels is processed in a second processing path including at least one FDN which applies the common late reverberation. Typically, the common late reverberation emulates collective macro attributes of late reverberation portions of at least some of the single-channel BRIRs. Other aspects are headphone virtualizers configured to perform any embodiment of the method.

METHOD FOR PROCESSING AN AUDIO SIGNAL, SIGNAL PROCESSING UNIT, BINAURAL RENDERER, AUDIO ENCODER AND AUDIO DECODER

A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is processed with an early part of the room impulse response separate from a late reverberation of the room impulse response, wherein the processing of the late reverberation has generating a scaled reverberated signal, the scaling being dependent on the audio signal. The processed early part of the audio signal and the scaled reverberated signal are combined. 1. A method for processing an audio signal in accordance with a room impulse response , the method comprising:separately processing the audio signal with an early part and a late reverberation of the room impulse response, wherein processing the late reverberation comprises generating a scaled reverberated signal, the scaling being dependent on the audio signal; andcombining the audio signal processed with the early part of the room impulse response and the scaled reverberated signal.2. The method of claim 1 , wherein the scaling is dependent on the condition of the one or more input channels of the audio signal.3. The method of claim 2 , wherein the condition of the one or more input channels of the audio signal comprises one or more of the number of input channels claim 2 , the number of active input channels and the activity in the input channel.4. The method of claim 1 , wherein the scaling is dependent on a predefined or calculated correlation measure of the audio signal.5. The method of claim 1 , wherein generating the scaled reverberated signal comprises applying a gain factor claim 1 , wherein the gain factor is determined based on the condition of the one or more input channels of the audio signal and/or based on the predefined or calculated correlation measure for the audio signal.6. The method of claim 5 , wherein generating the scaled reverberated signal comprises applying the gain factor before claim 5 , during or after processing the late reverberation of the audio signal.7. The ...

Method and apparatus for audio processing

A method and apparatus for introducing a time-varying time delay or phase shift 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.

SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD, AND PROGRAM

The present technology relates to a signal processing device, a signal processing method, and a program, which are capable of reproducing an acoustic field more appropriately in accordance with content. 1. A signal processing device comprising:an acquiring unit configured to acquire a multichannel audio signal obtained by performing sound collection through a microphone array;a spatial correction scheme selecting unit configured to select one spatial correction scheme from among a plurality of spatial correction schemes for correcting a spatial transfer characteristic on the basis of spatial correction information; anda spatial correction processing unit configured to perform a spatial correction process on the audio signal on the basis of a spatial transfer characteristic matrix of the selected spatial correction scheme.2. The signal processing device according to claim 1 ,wherein the spatial correction information is information indicating a priority of the spatial correction process.3. The signal processing device according to claim 1 ,wherein the spatial correction scheme selecting unit selects the spatial correction scheme on the basis of the spatial correction information and a number of speakers constituting a speaker array that outputs a sound on the basis of the audio signal.4. The signal processing device according to claim 1 ,wherein the spatial correction scheme selecting unit selects the spatial correction scheme on the basis of the spatial correction information and an operation capability of the signal processing device.5. The signal processing device according to claim 1 ,wherein the plurality of spatial correction schemes differ from each other in an operation amount of the spatial correction process.6. The signal processing device according to claim 1 ,wherein the spatial transfer characteristic matrix is obtained by extracting a part or a whole of a matrix indicating a spatial transfer characteristic of a space in which a sound based on the audio ...

METHOD AND SYSTEM FOR IMPLEMENTING A MODAL PROCESSOR

The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. 1. A method comprising:receiving a source signal; and processing the source signal in frequency subbands defined by a bank of filters, at least two of the filters having non-overlapping pass bands, transition bands, and stop bands, and with each frequency in the audio band appearing in either a single pass band, or a pair of transition bands,', 'wherein each mode filter of the plurality of mode filters is assigned to a single subband or a pair of subbands according to its respective mode frequency., 'applying artificial reverberation to the source signal by artificial reverberation to the source signal by processing the source signal in parallel using a plurality of mode filters having respective mode frequencies, the processing includes2. The method of claim 1 , further comprising:forming downsampled subband signals; andprocessing the downsampled subband signals with an associated down sampled mode filter.3. The method of claim 1 , further comprising:forming downsampled subbands through a process that includes heterodyning, low-pass filtering, and downsampling.4. The method of claim 3 , wherein the mode frequencies are adjusted according to the heterodyning frequency and downsampling factor.5. The method of claim 1 , wherein the mode filters have respective mode gains claim 1 , and wherein the mode gains are adjusted according to the characteristics of filters used to split the signal into subbands.6. The method of claim 5 , wherein the adjustments accommodate pass band ripple.7. The method of claim 1 ...

Reverberant Sound Adding Apparatus, Reverberant Sound Adding Method, and Reverberant Sound Adding Program

A reverberant sound adding apparatus includes a noise generator configured to generate a noise, an impulse noise generator configured to generate an impulse noise comprising an impulse sequence with random time intervals, an addition noise generator configured to generate an addition noise by adding the noise to the impulse noise, an impulse response generator configured to generate a modified impulse response by multiplying the addition noise by an amplitude characteristic of an impulse response that indicates acoustic characteristics of a space, and an impulse response convolver configured to convolve an input audio signal with the modified impulse response. 1. A reverberant sound adding apparatus , comprising:a noise generator configured to generate a noise;an impulse noise generator configured to generate an impulse noise comprising an impulse sequence with random time intervals;an addition noise generator configured to generate an addition noise by adding the noise to the impulse noise;an impulse response generator configured to generate a modified impulse response by multiplying the addition noise by an amplitude characteristic of an impulse response that indicates acoustic characteristics of a space; andan impulse response convolver configured to convolve an input audio signal with the modified impulse response.2. The reverberant sound adding apparatus according to claim 1 , wherein the addition noise generator comprises:a first multiplier configured to multiply the impulse noise by a first coefficient;a second multiplier configured to multiply the noise by a second coefficient; andan adder configured to generate the addition noise by adding an output of the first multiplier to an output of the second multiplier.3. The reverberant sound adding apparatus according to claim 1 , wherein the impulse response generator is configured to generate impulses at time intervals that are determined according to a prescribed probability density function to form the impulse ...

SYSTEM AND METHOD FOR SPATIAL PROCESSING OF SOUNDFIELD SIGNALS

A method for creating an output soundfield signal from an input soundfield signal, the method including the steps of: (a) forming at least one delayed signals from the input soundfield signal, (b) for each of the delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and (c) combining together the acoustically transformed delayed signals and the input soundfield signal to produce the output soundfield signal. 1. A method for creating an output soundfield signal from an input soundfield signal , the method including the steps of:(a) forming at least one delayed signal from said input soundfield signal,(b) for each of said delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and(c) combining together said acoustically transformed delayed signals and said input soundfield signal to produce said output soundfield signal.2. A method according to claim 1 , wherein the acoustic transformation process includes creating a direction of arrival of the respective delayed signal different from a direction of arrival of the input sound field claim 1 , relative to a listening position.3. A method according to claim 2 , wherein the direction of arrival of the respective delayed signal is created by applying a geometric transformation to the direction of arrival regarding the input sound field.4. A method as claimed in claim 1 , wherein the acoustic transformation process utilises a multi-channel matrix mixer.5. A method as claimed in claim 4 , wherein the multi-channel matrix mixer is formed by combining one or more spatial operations claim 4 , including a spatial rotation operation.6. A method as claimed in claim 4 , wherein the multi-channel matrix mixer is formed by combining one or more spatial operations claim 4 , including a spatial mirror operation.7. A method as claimed in any of claims 4 , wherein the multi-channel matrix mixer is formed by combining one or more ...

System and Method for Spatial Processing of Soundfield Signals

A method for creating an output soundfield signal from an input soundfield signal, the method including the steps of: (a) forming at least one delayed signals from the input soundfield signal, (b) for each of the delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and (c) combining together the acoustically transformed delayed signals and the input soundfield signal to produce the output soundfield signal. 1. A method for creating an output soundfield signal from an input soundfield signal , the method including the steps of:(a) forming at least one delayed signal from said input soundfield signal,(b) for each of said delayed signals, creating an acoustically transformed delayed signal, by an acoustic transformation process, and(c) combining together said acoustically transformed delayed signals and said input soundfield signal to produce said output soundfield signal.2. A method according to claim 1 , wherein the acoustic transformation process includes creating a direction of arrival of the respective delayed signal different from a direction of arrival of the input sound field claim 1 , relative to a listening position.3. A method according to claim 2 , wherein the direction of arrival of the respective delayed signal is created by applying a geometric transformation to the direction of arrival regarding the input sound field.4. A method as claimed in claim 1 , wherein the acoustic transformation process utilises a multi-channel matrix mixer.5. A method as claimed in claim 4 , wherein the multi-channel matrix mixer is formed by combining one or more spatial operations claim 4 , including a spatial mirror operation.6. A method as claimed in claim 4 , wherein the multi-channel matrix mixer is formed by combining one or more spatial operations claim 4 , including a directional gain operation.7. A method as claimed in claim 4 , wherein the multi-channel matrix mixer is formed by combining one or more spatial ...

LOCATION CLASSIFICATION FOR INTELLIGENT PERSONAL ASSISTANT

A personal assistant device configured to determine a location of the personal assistant device and apply settings specific to the location may include at least one speaker configured to emit a stimulus noise, at least one microphone configured to acquire a room sample based on the stimulus noise, the room sample including a sample room impulse response, and a memory configured to maintain a plurality of room profiles, each room profile including a room location, a stored room impulse response, and a room specific response. A processor may be configured to receive the room sample, compare the sample room impulse response with at least one of the stored room impulse responses, determine whether the sample room impulse response matches one of the stored room impulse responses, apply the room specific response associated with the room profile of the matched stored room impulse response. 1. A personal assistant device configured to determine a location of the personal assistant device and apply settings specific to the location , comprising:at least one speaker configured to emit a stimulus noise;at least one microphone configured to acquire a room sample based on the stimulus noise, the room sample including a sample room impulse response;a memory configured to maintain a plurality of room profiles, each room profile including a room location, a stored room impulse response, and a room specific response; receive the room sample;', 'compare the sample room impulse response with at least one of the stored room impulse responses;', 'determine whether the sample room impulse response matches one of the stored room impulse responses; and', 'apply the room specific response associated with the room profile of the matched stored room impulse response., 'a processor configured to2. The device of claim 1 , wherein the room sample includes a logarithmic frequency sweep of a predetermined amount of time.3. The device of claim 1 , wherein the sample room impulse response includes ...

VEHICLE AND OPERATION METHOD THEREOF

A device can include a microphone, a memory configured to store a sound source table including correspondence relationships among boarding items, virtual sound items and filter set values, and a processor configured to receive a sound signal through the microphone and determine a first filter set value matching characteristics of the received sound signal using the sound source table. 1. A device comprising:a microphone;a memory configured to store a sound source table including correspondence relationships among boarding items, virtual sound items and filter set values; and receive a sound signal through the microphone, and', 'determine a first filter set value matching characteristics of the received sound signal using the sound source table., 'a processor configured to2. The device of claim 1 , wherein the processor is configured to:extract characteristics of the sound signal and characteristics of a passenger signal obtained by removing the sound signal from a reference signal,determine boarding items matching the characteristics of the sound signal and the characteristics of the passenger signal, anddetermine the first filter set value matching the determined boarding items using the sound field table.3. The device of claim 2 , further comprising a camera claim 2 ,wherein the processor is configured to:acquire second boarding information including the number passengers and the seat position of each passenger from an image captured through the camera,compare first boarding information and second boarding information including boarding items matching the characteristics of the sound signal and the characteristics of the passenger signal, anddetermine a second filter set value matching the second boarding information using the sound field table, when the first boarding information and the second boarding information are different.4. The device of claim 3 , wherein the processor is configured to change the first filter set value to the second filter set value using ...

METHOD AND SYSTEM FOR IMPLEMENTING A MODAL PROCESSOR

The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time. 1. A method comprising:receiving a source signal; andapplying artificial reverberation to the source signal by processing the source signal in parallel using a plurality of mode filters having respective mode frequencies,wherein each mode filter of the plurality of mode filters is assigned to a single subband or a pair of subbands according to its respective mode frequency, andwherein a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.2. The method of claim 1 , wherein the feature is mode filter amplitude.3. The method of claim 1 , wherein the feature is mode filter phase.4. The method of claim 1 , wherein the aspect is an overall or weighted residual square error.5. The method of claim 1 , wherein the processing includes processing the source signal in frequency subbands defined by a bank of filters claim 1 , at least two of the filters having non-overlapping pass bands or transition bands claim 1 , and with each frequency in the audio band substantially appearing in either a single pass band or a pair of transition bands.6. A system for implementing an approximation to a desired filter comprisinga modal process; anda convolutional process combined in parallel with the modal process,wherein the modal process comprises the parallel combination of a plurality of mode ...

SIGNAL PROCESSING DEVICE, METHOD, AND PROGRAM

The present technology relates to a signal processing device, method, and program that can improve encoding efficiency. 1. A signal processing device comprising:an acquisition unit that acquires reverb information including at least one of space reverb information specific to a space around an audio object or object reverb information specific to the audio object and an audio object signal of the audio object; anda reverb processing unit that generates a signal of a reverb component of the audio object on a basis of the reverb information and the audio object signal.2. The signal processing device according to claim 1 , wherein the space reverb information is acquired at a lower frequency than the object reverb information.3. The signal processing device according to claim 1 , wherein in a case where identification information indicating past reverb information is acquired by the acquisition unit claim 1 , the reverb processing unit generates a signal of the reverb component on a basis of the reverb information indicated by the identification information and the audio object signal.4. The signal processing device according to claim 3 , wherein the identification information is information indicating the object reverb information claim 3 , andthe reverb processing unit generates a signal of the reverb component on a basis of the object reverb information indicated by the identification information, the space reverb information, and the audio object signal.5. The signal processing device according to claim 1 , wherein the object reverb information is information depending on a position of the audio object.6. The signal processing device according to claim 1 , wherein the reverb processing unitgenerates a signal of the reverb component specific to the space on a basis of the space reverb information and the audio object signal, andgenerates a signal of the reverb component specific to the audio object on a basis of the object reverb information and the audio object ...

VEHICLE AND OPERATION METHOD THEREOF

A vehicle according to an embodiment of the present invention may comprise: a microphone; a plurality of speakers; a memory for storing a sound field table including corresponding relationship between filter set values, virtual sound items, and boarding items; and a processor for receiving a sound signal via the microphone, determining a first filter set value matched to characteristics of the received sound signal, by using the sound field table, and controlling outputs of the plurality of speakers so as to output sound according to the determined first set value. 1. A vehicle comprising:a microphone;a plurality of speakers;a memory configured to store a sound source table including correspondence relationships among boarding items, virtual sound items and filter set values; anda processor configured to receive a sound signal through the microphone, determine a first filter set value matching characteristics of the received sound signal using the sound source table, and control output of the plurality of speakers to output sound according to the determined first filter set value.2. The vehicle of claim 1 , wherein the processor is configured to:extract characteristics of the sound signal and characteristics of a passenger signal obtained by removing the sound signal from a reference signal,determine boarding items matching the characteristics of the sound signal and the characteristics of the passenger signal, anddetermine the first filter set value matching the determined boarding items using the sound field table.3. The vehicle of claim 1 ,wherein the boarding items include the number of passengers and a seat position of each passenger, andwherein the virtual sound signals include an early decay time, a first reverberation time and a second reverberation signal of each of a virtual sound signal and a passenger signal obtained by removing the virtual sound signal from a reference signal.4. The vehicle of claim 2 , further comprising a camera claim 2 ,wherein the ...

SOUND OUTPUT DEVICE, SOUND OUTPUT METHOD, AND SOUND OUTPUT SYSTEM

According to the present disclosure, a sound output device includes a sound acquisition part that acquires a sound signal generated from an ambient sound, a reverb process part that performs a reverb process on the sound signal, and a sound output part that outputs a sound generated from the sound signal subjected to the reverb process, to a vicinity of an ear of a listener. This configuration allows a listener to hear sound acquired in real time to which desired reverberation is added. 1. A first sound output device , comprising:a microphone;first circuitry configured to acquire a first sound environment information that indicates a first sound environment around the first sound output device;a wireless communication part configured to acquire, from a second sound output device, a second sound environment information that indicates a second sound environment around the second sound output device, wherein a location of the second sound output device is remote from the first sound output device; acquire, by the microphone, a first sound signal generated from an ambient sound around the first sound output device;', 'reverberate the first sound signal based on the first sound environment information and the second sound environment information; and', 'generate a sound based on the reverberated first sound signal; and, 'a sound generating portion comprising second circuitry configured toa sound guiding portion configured to propagate the generated sound from a sound input hole of the sound guiding portion to a sound output hole of the sound guiding portion.2. The first sound output device according to claim 1 , further comprising a supporting portion having a hollow structure claim 1 , whereinthe supporting portion comprises a support member and an opening portion in the hollow structure, andthe opening portion is different from the support member.3. The first sound output device according to claim 2 , wherein the supporting portion has a ring-shaped structure.4. The ...

METHOD FOR PROCESSING AN AUDIO SIGNAL, SIGNAL PROCESSING UNIT, BINAURAL RENDERER, AUDIO ENCODER AND AUDIO DECODER

A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is processed with an early part of the room impulse response separate from a late reverberation of the room impulse response, wherein the processing of the late reverberation has generating a scaled reverberated signal, the scaling being dependent on the audio signal. The processed early part of the audio signal and the scaled reverberated signal are combined. 1. A method for processing an audio signal in accordance with a room impulse response , the method comprising:separately processing the audio signal with an early part and a late reverberation of the room impulse response, wherein processing the late reverberation comprises generating a scaled reverberated signal; andcombining the audio signal processed with the early part of the room impulse response and the scaled reverberated signal, setting a gain factor according to a predefined correlation measure of the audio signal comprising a fixed value determined empirically on the basis of an analysis of a plurality of audio signals, and applying the gain factor, or', 'calculating a correlation measure of the audio signal using a correlation analysis of the audio signal and the gain factor, and applying the gain factor., 'wherein generating a scaled reverberated signal comprises'}2. The method of claim 1 , wherein the scaling is dependent on a condition of one or more input channels of the audio signal claim 1 , wherein the condition of the one or more input channels of the audio signal comprises one or more of the number of input channels claim 1 , the number of active input channels claim 1 , and an activity in the one or more input channel.3. The method of claim 2 , wherein the gain factor is determined based on the condition of the one or more input channels of the audio signal.4. The method of claim 3 , wherein generating the scaled reverberated signal comprises applying the gain factor before ...

Apparatus and method for generating virtual venues for a listening room

In at least one embodiment, an apparatus for generating virtual venues in a listening room in a vehicle is provided. The apparatus includes a vehicle audio controller configured to receive an incoming audio signal including entertainment data from an audio source for playback and to receive at least one captured audio signal from at least one microphone positioned in the listening room of the vehicle. The vehicle audio controller is further configured to receive a control signal indicative of a desired venue to playback the entertainment data and to adjust reverberation of at least one of the incoming audio signal and the at least one captured audio signal to playback the entertainment data in the desired venue.

SPATIAL AUDIO FOR INTERACTIVE AUDIO ENVIRONMENTS

Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment. The respective second intermediate audio signal is determined based on a location of the respective sound source, and further based on an acoustic property of the virtual environment. The respective second intermediate audio signal is associated with a second bus. The output audio signal is presented to the listener via the first bus and the second bus. 1determining, based on a location of a sound source in a virtual environment, a first intermediate audio signal corresponding to an input audio signal;associating the first intermediate audio signal with a first bus;determining, based on a location of the sound source and further based on an acoustic property of the virtual environment, a second intermediate audio signal, the second intermediate audio signal corresponding to a reverberation of the input audio signal in the virtual environment; andassociating the second intermediate audio signal with a second bus; andpresenting, via the first bus and the second bus, the output audio signal to a listener.. A method comprising: This application is a continuation of U.S. patent application Ser. No. 16/905,855, filed on Jun. 18, 2020, which is a ...

Sound signal processing method and sound signal processing device

A sound signal processing method includes: receiving a line-inputted sound signal; controlling a volume of the line-inputted sound signal; and generating an early reflected sound control signal using the line-inputted sound signal having the controlled volume.

SOUND SIGNAL PROCESSING METHOD AND SOUND SIGNAL PROCESSING DEVICE

A sound signal processing method includes: obtaining a plurality of sound signals respectively collected by a plurality of microphones arranged in a space; adjusting respective levels of the plurality of sound signals in accordance with respective positions of the plurality of microphones; mixing the plurality of sound signals having the adjusted respective levels to thereby obtain a mixed signal; and generating a reflected sound by using the obtained mixed signal. 1. A sound signal processing method comprising:obtaining a plurality of sound signals respectively collected by a plurality of microphones arranged in a space;adjusting respective levels of the plurality of sound signals in accordance with respective positions of the plurality of microphones;mixing the plurality of sound signals having the adjusted respective levels to thereby obtain a mixed signal; andgenerating a reflected sound by using the obtained mixed signal.2. The sound signal processing method according to claim 1 , wherein adjusting the respective levels of the plurality of sound signals includes adjusting each of the respective levels of the plurality of sound signals in accordance with a distance from each of the respective positions of the plurality of microphones to a speaker that outputs the reflected sound.3. The sound signal processing method according to claim 2 , wherein each of the respective levels of the plurality of sound signals are adjusted by setting a gain for each of the plurality of sound signals to be higher in ascending order of the distance from each of the respective positions of the plurality of microphones to the speaker that outputs the reflected sound.4. The sound signal processing method according to claim 1 , further comprisingadjusting a delay time of the reflected sound in accordance with a distance from each of the respective positions of the plurality of microphones to a speaker that outputs the reflected sound.5. The sound signal processing method according to ...

APPARATUS AND METHOD FOR GENERATING OUTPUT SIGNALS BASED ON AN AUDIO SOURCE SIGNAL, SOUND REPRODUCTION SYSTEM AND LOUDSPEAKER SIGNAL

An apparatus for generating a first multitude of output signals based on at least one audio source signal having a delay network and a feedback processor. The delay network includes a second multitude of delay paths, each delay path having a delay line and an attenuation filter. Each delay line is configured for delaying delay line input signals and for combining the at least one audio source signal and a reverberated audio signal to obtain a combined signal, wherein the attenuation filter of a delay path is configured for filtering the combined signal from the delay line of the delay path to obtain an output signal. The first multitude of output signals includes the output signal. The feedback processor is configured for reverberating the first multitude of output signals to obtain a third multitude of the reverberated audio signals including the reverberated audio signal. 1. Apparatus for generating a first multitude of output signals based on at least one audio source signal , the apparatus comprising:a delay network comprising a second multitude of delay paths each delay path comprising a delay line and an attenuation filter, each delay line being configured for delaying delay line input signals and for combining the at least one audio source signal and a reverberated audio signal to acquire a combined signal, wherein the attenuation filter of a delay path is configured for filtering the combined signal from the delay line of the delay path to acquire an output signal, wherein the first multitude of output signals comprises the output signal; anda feedback processor configured for reverberating the first multitude of output signals to acquire a third multitude of the reverberated audio signals comprising the reverberated audio signal;wherein the combined signal comprises an audio source signal portion and a reverberated signal portion and wherein the delay line comprises a sixth multitude of input taps being configured for receiving the audio source signal or a ...

INFORMATION PROCESSING APPARATUS, INFORMATION PROCESSING SYSTEM, INFORMATION PROCESSING METHOD, AND PROGRAM

There is provided an information processing apparatus including an effect control unit configured to control an echo effect that is to be added to an output, a sound output control unit configured to control a sound output according to occurrence of an event in a virtual space, and a haptic output control unit configured to control a haptic output to which the echo effect controlled by the effect control unit is added according to the occurrence of the event. 1. An information processing apparatus comprising:an effect control unit configured to control an echo effect that is to be added to an output;a sound output control unit configured to control a sound output according to occurrence of an event in a virtual space; anda haptic output control unit configured to control a haptic output to which the echo effect controlled by the effect control unit is added according to the occurrence of the event.2. The information processing apparatus according to claim 1 ,wherein the effect control unit controls the echo effect on the basis of a delay time from an output start instruction on the haptic output to occurrence of the output.3. The information processing apparatus according to claim 2 ,wherein the effect control unit increases or decreases an effect amount of the echo effect according to the delay time.4. The information processing apparatus according to claim 3 ,wherein the haptic output control unit decreases the haptic output when the delay time exceeds a certain threshold.5. The information processing apparatus according to claim 2 ,wherein the effect control unit decides presence or absence of the echo effect on the basis of the delay time and a certain threshold.6. The information processing apparatus according to claim 1 ,wherein the effect control unit decides an effect amount of the echo effect on the basis of a user's operation.7. The information processing apparatus according to claim 1 ,wherein the haptic output control unit decreases the haptic output ...

ACOUSTIC APPARATUS AND VIBRATION TRANSMISSION METHOD

Provided is an acoustic apparatus capable of causing a sensation that a sound is produced in a larger space than reality, without requiring an excessive cost and time-consuming installation. The sound collected by a microphone is converted to an electrical signal, and a reflected sound component is extracted by a signal processing portion. The reflected sound component thus extracted is amplified by an amplifier, and the reflected sound component thus amplified vibrates transducers attached to a back-side board surface of each acoustic structure, whereby reflected sound is emitted from all faces including front-side board surface and the back-side board surface. A player can thus naturally perceive as if the space where he stays is expanded. 1. An acoustic apparatus comprising:a signal acquisition portion that acquires a signal that is at least one of a sound signal and a vibration signal generated by a sound source;a signal processing portion that subjects the signal acquired by the signal acquisition portion to a predetermined signal processing;an acoustic structure having a plate-like shape and comprising thereinside at least one cavity that extends in one direction; anda vibration generation portion that is attached to the acoustic structure,wherein the vibration generation portion generates vibration corresponding to the signal acquired by the signal acquisition portion and transmits the vibration generated to the acoustic structure.2. The acoustic apparatus according to claim 1 , wherein the acoustic structure comprises at least one opening that communicates the cavity with an outer space.3. The acoustic apparatus according to claim 1 , wherein the vibration generation portion transmits the vibration to a position corresponding to a node of a vibration mode of the acoustic structure claim 1 , on an outer face of the acoustic structure.4. The acoustic apparatus according to claim 1 , wherein the vibration generation portion transmits the vibration to a position ...

Audio Signal Rendering

An apparatus for audio signal rendering, the apparatus including at least one processor configured to: receive at least one microphone audio signal captured by at least one microphone within a capture environment; receive at least one projection audio signal, wherein the at least one projection audio signal is a room-impulse-response filtered at least one microphone audio signal within the capture environment; receive at least one residual audio signal, wherein the at least one residual audio signal is a result of removing the at least one projection audio signal from at least one audio signal captured by a microphone array within the capture environment; and generate at least two volumetric audio signals based on the at least one microphone audio signal, the at least one projection audio signal and the at least one residual audio signal.

Method for Sound Effect Compensation, Non-Transitory Computer-Readable Storage Medium, and Terminal Device

A method and an apparatus for sound effect compensation, a non-transitory computer-readable storage medium, and a terminal device are provided. The method includes the following. An acoustic transmission function of an audio processing device in an ear canal of a user is determined according to an audio signal currently output from the audio processing device. Compensate for the audio signal according to the acoustic transmission function. 1. A method for sound effect compensation , comprising:determining an acoustic transmission function of an audio processing device in an ear canal of a user according to an audio signal output from the audio processing device; andcompensating for the audio signal according to the acoustic transmission function.2. The method of claim 1 , wherein the compensating for the audio signal according to the acoustic transmission function comprises:determining a monitoring frequency-response curve of the audio processing device according to the acoustic transmission function; andcompensating for the audio signal by adjusting the monitoring frequency-response curve according to a standard frequency-response curve.3. The method of claim 1 , wherein the determining an acoustic transmission function of an audio processing device in an ear canal of a user according to an audio signal output from the audio processing device comprises:acquiring the audio signal output from the audio processing device;recording an acoustic echo signal formed by reflection and vibration of the audio signal through the ear canal; anddetermining the acoustic transmission function of the audio processing device in the ear canal of the user according to the audio signal and the acoustic echo signal.4. The method of claim 2 , wherein the determining a monitoring frequency-response curve of the audio processing device according to the acoustic transmission function comprises:acquiring a corresponding relationship among acoustic echo signals, acoustic transmission ...

REVERBERATOR AND METHOD FOR REVERBERATING AN AUDIO SIGNAL

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.

SPATIAL AUDIO FOR INTERACTIVE AUDIO ENVIRONMENTS

Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment. The respective second intermediate audio signal is determined based on a location of the respective sound source, and further based on an acoustic property of the virtual environment. The respective second intermediate audio signal is associated with a second bus. The output audio signal is presented to the listener via the first bus and the second bus. 1. A method comprising:determining, based on a location of a sound source in a virtual environment, a first intermediate audio signal corresponding to an input audio signal;associating the first intermediate audio signal with a first bus;determining, based on a location of the sound source and further based on an acoustic property of the virtual environment, a second intermediate audio signal, the second intermediate audio signal corresponding to a reverberation of the input audio signal in the virtual environment; andassociating the second intermediate audio signal with a second bus; andpresenting, via the first bus and the second bus, the output audio signal to a listener.2. The method of claim 1 , wherein the acoustic property of the virtual environment is determined via one or more ...

METHOD FOR PROCESSING AN AUDIO SIGNAL IN ACCORDANCE WITH A ROOM IMPULSE RESPONSE, SIGNAL PROCESSING UNIT, AUDIO ENCODER, AUDIO DECODER, AND BINAURAL RENDERER

A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response. 1. A method for processing an audio signal in accordance with a room impulse response , the method comprising:separately processing the audio signal with an early part of the room impulse response and with a late reverberation of the room impulse response or a synthetic reverberation, wherein processing the audio signal with the early part of the room impulse response comprises a convolution of the audio signal with a direct sound and early reflections of the room impulse response; andcombining the audio signal processed with the early part of the room impulse response and the audio signal processed with the late reverberation of the room impulse response or with the synthetic reverberation,determining a transition from the early part to the late reverberation in the room impulse response as a time when a correlation measure reaches a threshold,wherein the correlation measure describes with regard to the room impulse response a similarity of a decay in acoustic energy comprising an initial state and of the decay in acoustic energy starting at a point in time, said point in time following the initial state over a predefined frequency range,wherein the threshold is set dependent on the correlation measure for said point in time, said point in time being a time of a selected one of the early reflections in the early part of the room impulse response andwherein the selected one ...

AUDIO DEVICE AND PROGRAM

A device is provided which adds a desired delay (echo) effect while suppressing increase in the cost with a simple structure. A CPU of an audio device activates a plurality of media players A, B, C, . . . for reproducing an audio file. The audio file is reproduced as an original sound by one of the plurality of media players, and a delay sound with respect to the original sound of the audio file is reproduced by another media player, and is added to the original sound. 1. An audio device comprising:a storage unit that stores a plurality of player applications for reproducing an audio file; anda controller that activates the plurality of player applications, that reproduces the audio file as an original sound by one of the plurality of player applications, that simultaneously reproduces a delay sound with respect to the original sound of the audio file by each of one or more of the player applications other than the one of the plurality of player applications, and that adds the delay sound reproduced by each of the one or more of the player applications to the original sound.2. The audio device according to claim 1 , further comprising:a setter that sets a parameter of a delay effect for each of the one or more of the player applications, whereinthe controller reproduces the delay sound by each of the one or more player applications with a delay time and an output level according to the parameter which is set by the setter for each of the one or more of the player applications.3. A non-transitory recording medium which stores a program which claim 1 , when executed claim 1 , causes a processor of a computer to:activate a plurality of player applications for reproducing an audio file; andreproduce the audio file as an original sound by one of the plurality of player applications, simultaneously reproduce a delay sound with respect to the original sound of the audio file by each of one or more of the player applications other than the one of the plurality of player ...

SOUND EFFECT PRODUCING APPARATUS, METHOD OF PRODUCING SOUND EFFECT AND PROGRAM THEREFOR

A sound effect producing apparatus includes an input portion, a memory portion, a pseudo sound reflection producing portion, and an effect provision portion. The input portion performs inputting of an audio signal. The memory portion stores sound effect information that includes production information for producing a pseudo sound reflection corresponding to a sound reflection generated in a predetermined acoustic space and sound source position information showing a sound source position of the pseudo sound reflection. The pseudo sound reflection producing portion produces a pseudo sound reflection based on the production information. The effect provision portion performs a process of localizing the pseudo sound reflection using a predetermined direction as a reference, based on the audio signal and the sound source position information. 1. A sound effect producing apparatus comprising:an input portion that performs inputting of an audio signal;a memory portion that stores sound effect information which includes production information for producing a pseudo sound reflection corresponding to a sound reflection generated in a predetermined acoustic space and sound source position information showing a sound source position of the pseudo sound reflection;a pseudo sound reflection producing portion that produces a pseudo sound reflection based on the production information; andan effect provision portion that performs a process of localizing the pseudo sound reflection using a predetermined direction as a reference, based on the sound source position information.2. The sound effect producing apparatus according to claim 1 , whereinthe effect provision portion performs the process of localizing the pseudo sound reflection using a predetermined direction as a reference, by changing the sound source position information stored in the memory portion.3. The sound effect producing apparatus according to claim 1 , whereinthe memory portion stores a plurality of pieces of the ...

SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD

A signal processing apparatus includes: an obtaining portion configured to obtain an impulse response in a semi-open state in which, among a plurality of acoustic feedback systems, at least one acoustic feedback system is open and at least one acoustic feedback system is closed; and a calculating portion configured to calculate a gain correction amount based on the impulse response that the obtaining portion has obtained and to output a calculated gain correction amount. 1. A signal processing apparatus comprising:an obtaining portion configured to obtain an impulse response in a semi-open state in which, among a plurality of acoustic feedback systems, at least one acoustic feedback system is open and at least one acoustic feedback system is closed; anda calculating portion configured to calculate a gain correction amount based on the impulse response that the obtaining portion has obtained and to output a calculated gain correction amount.2. The signal processing apparatus according to claim 1 , wherein the calculating portion is configured to calculate a target frequency of coloration and calculate the gain correction amount so as to reduce a gain of the target frequency.3. The signal processing apparatus according to claim 2 , wherein the calculating portion is configured to set a frequency that shows one or more peaks to the target frequency.4. The signal processing apparatus according to claim 3 , whereinthe one or more peaks include a first peak of which a level is relatively large, and a second peak; andthe calculating portion, when calculating the target frequency, excludes the second peak in a case in which the first peak is set as a standard and a frequency of the second peak is in a predetermined band around a frequency of the first peak.5. The signal processing apparatus according to claim 3 , whereinthe calculating portion is configured to calculate a difference between a moving average on a frequency axis of the impulse response and each of the one or ...

Resonance Signal Generating Method, Resonance Signal Generating Device, Electronic Musical Apparatus and Non-Transitory Computer Readable Medium

A resonance signal generating method includes generating a first resonance signal of a specific pitch circulating through first loop processing by inputting an excitation signal to the first loop processing including first delay that delays the signal by a time corresponding to the specific pitch and first attenuation that attenuates the signal, generating a second resonance signal of the specific pitch circulating through second loop processing by the second loop processing including second delay that delays the signal by a time corresponding to the specific pitch and second attenuation that attenuates the signal, generating an attenuation change signal by adding the first resonance signal and the second resonance signal to each other, attenuating and inverting the added first and second resonance signals, and inputting the generated attenuation change signal to the first loop processing and the second loop processing respectively, and outputting the first resonance signal. 1. A resonance signal generating method comprising:generating a first resonance signal of a specific pitch circulating through first loop processing by inputting an excitation signal to the first loop processing including first delay that delays the signal by a time corresponding to the specific pitch and first attenuation that attenuates the signal;generating a second resonance signal of the specific pitch circulating through second loop processing by the second loop processing including second delay that delays the signal by a time corresponding to the specific pitch and second attenuation that attenuates the signal;generating an attenuation change signal by adding the first resonance signal and the second resonance signal to each other, attenuating and inverting the added first and second resonance signals, and inputting the generated attenuation change signal to the first loop processing and the second loop processing; andoutputting the first resonance signal.2. The resonance signal ...

ACTIVE REVERBERATION AUGMENTATION

A method for using a loudspeaker array that is housed in a loudspeaker cabinet to present audio content to a listener in a room includes receiving (1) an audio channel that includes audio content and (2) acoustical characteristics of the room. The method also produces (1) a first beamformer input signal from the audio channel and (2) a second beamformer input signal and a third beamformer input signal by decorrelating the audio channel and adjusting the audio channel in accordance with the acoustical characteristics of the room. The second and third beamformer input signals are different de-correlated versions of the audio channel. The method also generates driver signals from the first, second, and third beamformer input signals to drive the loudspeaker array to produce a main beam, a first ambient beam, and a second ambient beam, respectively. Other embodiments are also described and claimed. 1. A method for using a loudspeaker array that is housed in a loudspeaker cabinet to present audio content to a listener in a room , the method comprising:receiving, by a rendering signal processor, (1) an audio channel that includes audio content that is to be converted into sound by the loudspeaker array housed in the loudspeaker cabinet and (2) acoustical characteristics of the room;producing, by the rendering signal processor, a first beamformer input signal from the audio channel;processing, by the rendering signal processor, the audio channel, and adjusting the audio channel in accordance with the acoustical characteristics of the room, to produce a processed and adjusted audio channel as a second beamformer input signal;processing, by the rendering signal processor, the audio channel, and adjusting the audio channel in accordance with the acoustical characteristics of the room, to produce a further processed adjusted audio channel as a third beamformer input signal, wherein the second and third beamformer input signals are different processed versions of the audio ...

REVERBERATOR AND METHOD FOR REVERBERATING AN AUDIO SIGNAL

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. 1. Reverberator for reverberating an audio signal , comprising:a feedback delay loop processor for delaying at least two different frequency subband signals representing the audio signal by different loop delays to acquire reverberated frequency subband signals,wherein the feedback delay loop processor comprises for a first frequency subband signal of the at least two frequency subband signals a first delay line comprising a first plurality of delay line taps providing signals delayed by different tap delays, a first feedback loop connected to the first delay line and a first combiner for combining signals output by the first plurality of delay line taps, andwherein the feedback delay loop processor comprises for a second different frequency subband signal of the at least two frequency subband signals a second delay line comprising a second plurality of delay line taps providing signals delayed by different tap delays, a second feedback loop connected to the second delay line and a second combiner for combining signals output by the second plurality of delay line taps.2. The reverberator according to claim 1 , further comprising an output processor for processing the reverberated frequency subband signals to aqcuire a reverberated audio signal.3. The reverberator according to claim 1 , wherein the output processor performs mixing the at least two frequency subband signals and the corresponding reverberated frequency subband signals to acquire mixed signals and to combine the mixed signals claim 1 , or combining the reverberated frequency subband signals to acquire the reverberated audio signal comprising a combined bandwidth.4. The reverberator according to claim 1 , wherein the feedback delay loop processor ...

SPATIAL AUDIO FOR INTERACTIVE AUDIO ENVIRONMENTS

Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment. The respective second intermediate audio signal is determined based on a location of the respective sound source, and further based on an acoustic property of the virtual environment. The respective second intermediate audio signal is associated with a second bus. The output audio signal is presented to the listener via the first bus and the second bus. 1. A method of presenting an output audio signal to a listener located at a first location in a virtual environment , the method comprising:receiving an input audio signal; determining, based on a location of the respective sound source in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal;', 'associating the respective first intermediate audio signal with a first bus;', 'determining, based on a location of the respective sound source and further based on an acoustic property of the virtual environment, a respective second intermediate audio signal, the respective second intermediate audio signal corresponding to a reverberation of the input audio signal in the virtual environment; and', 'associating the respective second ...

Method for processing an audio signal in accordance with a room impulse response, signal processing unit, audio encoder, audio decoder, and binaural renderer

A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response.

REVERBERATION ADDING APPARATUS, REVERBERATION ADDING METHOD, AND REVERBERATION ADDING PROGRAM

A reverberation adding apparatus includes: a plurality of paths that constitutes an output channel or a plurality of output channels; and a convolution operation unit that convolves an impulse response for each of the paths, in which the impulse response is formed by combining a plurality of reverberation pattern blocks in a time axis direction, the reverberation adding apparatus using the reverberation pattern blocks common to the plurality of convolution operation units. 1. A reverberation adding apparatus , comprising:a plurality of paths that constitutes an output channel or a plurality of output channels; anda convolution operation unit that convolves an impulse response for each of the paths, whereinthe impulse response is formed by combining a plurality of reverberation pattern blocks in a time axis direction, the reverberation adding apparatus using the reverberation pattern blocks common to the plurality of convolution operation units.2. The reverberation adding apparatus according to claim 1 , whereinthe reverberation pattern blocks common to the plurality of convolution operation units are different in use time position.3. The reverberation adding apparatus according to claim 2 , whereinthe reverberation pattern blocks common to the plurality of convolution operation units are multiplied with a coefficient according to the use time position.4. The reverberation adding apparatus according to claim 1 , whereinthe impulse response to be used in each of the plurality of convolution operation units arranged on the paths that constitute the identical output channel has a common permutation of the reverberation pattern blocks.5. The reverberation adding apparatus according to claim 1 , whereinthe plurality of same reverberation pattern blocks are not used in the impulse response.6. The reverberation adding apparatus according to claim 1 , whereinthe same reverberation pattern blocks are not temporally adjacent in the impulse response.7. The reverberation adding ...

Устройство, способ и программа для обработки сигнала

РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2020 112 483 A (51) МПК G10K 15/12 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ЗАЯВКА НА ИЗОБРЕТЕНИЕ (21)(22) Заявка: 2020112483, 05.10.2018 (71) Заявитель(и): СОНИ КОРПОРЕЙШН (JP) Приоритет(ы): (30) Конвенционный приоритет: 20.10.2017 JP 2017-203877 (85) Дата начала рассмотрения заявки PCT на национальной фазе: 27.03.2020 R U (43) Дата публикации заявки: 27.09.2021 Бюл. № 27 (72) Автор(ы): ХОНМА Хироюки (JP), ЦУДЗИ Минору (JP), ТИНЕН Тору (JP) (86) Заявка PCT: (87) Публикация заявки PCT: WO 2019/078035 (25.04.2019) R U (54) УСТРОЙСТВО, СПОСОБ И ПРОГРАММА ДЛЯ ОБРАБОТКИ СИГНАЛА (57) Формула изобретения 1. Устройство для обработки сигнала, содержащее: модуль сбора данных, который получает реверберационную информацию, содержащую информацию по меньшей мере одного вида – пространственную реверберационную информацию, специфичную для пространства вокруг аудиообъекта, и/или объектную реверберационную информацию, специфичную для самого этого аудиообъекта, и сигнал этого аудиообъекта; и модуль реверберационной обработки, генерирующий сигнал реверберационной составляющей аудиообъекта на основе реверберационной информации и сигнала этого аудиообъекта. 2. Устройство для обработки сигнала по п. 1, отличающееся тем, что пространственную реверберационную информацию получают с меньшей частотой, чем объектную реверберационную информацию. 3. Устройство для обработки сигнала по п. 1, отличающееся тем, что, если модулем сбора данных получена идентификационная информация, указывающая прошлую реверберационную информацию, модуль реверберационной обработки генерирует сигнал реверберационной составляющей на основе реверберационной информации, указанной посредством идентификационной информации, и сигнала аудиообъекта. 4. Устройство для обработки сигнала по п. 3, отличающееся тем, что идентификационная информация представляет собой информацию, указывающую Стр.: 1 A 2 0 2 0 1 1 2 4 8 3 A Адрес для переписки: 101000, ул. Мясницкая, ...

Patent RU2020112483A3

ВИ“? 2020112483`” АЗ Дата публикации: 21.04.2022 Форма № 18 ИЗПМ-2011 Федеральная служба по интеллектуальной собственности Федеральное государственное бюджетное учреждение ж 5 «Федеральный институт промышленной собственности» (ФИПС) ОТЧЕТ О ПОИСКЕ 1. . ИДЕНТИФИКАЦИЯ ЗАЯВКИ Регистрационный номер Дата подачи 2020112483/28(021118) 05.10.2018 РСТ/]Р2018/037330 05.10.2018 Приоритет установлен по дате: [ ] подачи заявки [ ] поступления дополнительных материалов от к ранее поданной заявке № [ ] приоритета по первоначальной заявке № из которой данная заявка выделена [ ] подачи первоначальной заявки № из которой данная заявка выделена [ ] подачи ранее поданной заявки № [Х] подачи первой(ых) заявки(ок) в государстве-участнике Парижской конвенции (31) Номер первой(ых) заявки(ок) (32) Дата подачи первой(ых) заявки(ок) (33) Код страны 1. 2017-203877 20.10.2017 ]Р Название изобретения (полезной модели): [Х] - как заявлено; [ ] - уточненное (см. Примечания) УСТРОЙСТВО, СПОСОБ И ПРОГРАММА ДЛЯ ОБРАБОТКИ СИГНАЛА Заявитель: СОНИ КОРПОРЕЙШН, ]Р 2. ЕДИНСТВО ИЗОБРЕТЕНИЯ [Х] соблюдено [ ] не соблюдено. Пояснения: см. Примечания 3. ФОРМУЛА ИЗОБРЕТЕНИЯ: [Х] приняты во внимание все пункты см. п см. Примечания [ ] приняты во внимание следующие пункты: р [ ] принята во внимание измененная формула изобретения (см. Примечания) 4. КЛАССИФИКАЦИЯ ОБЪЕКТА ИЗОБРЕТЕНИЯ (ПОЛЕЗНОЙ МОДЕЛИ) (Указываются индексы МПК и индикатор текущей версии) СТОК 15/12 (2006.01) С.10Г. 19/008 (2013.01) НО45 5/02 (2006.01) 5. ОБЛАСТЬ ПОИСКА 5.1 Проверенный минимум документации РСТ (указывается индексами МПК) СТОК 15/12 (2006.01)Г С1ОГ, 19/008 (2013.00 Н04$ 5/02 (2006.01)1 5.2 Другая проверенная документация в той мере, в какой она включена в поисковые подборки: 5.3 Электронные базы данных, использованные при поиске (название базы, и если, возможно, поисковые термины): ВУРАТЕМТЬЗ, ОУУРТ, Е-ГлЬгагу, ЕАРАТТЪ, ЕВЗСО, Езрасепе боозе, Соое Ржепб, /- Р]а{ Рав, КТРКЗХ, Г.ех15Мех15, РАТЕМТСОРЕ, Ра еагсь, Оцез{е]-Отфи, КОРТО, ...

Audio system and method for operation thereof

Изобретение относится к аудиосистеме, в частности к виртуальному пространственному воспроизведению аудиосигналов. Технический результат - обеспечение полного восприятия пространственного звука на основе бинауральных сигналов. Аудиосистема содержит приемник для приема аудиосигнала, такого как аудиообъект или сигнал канала пространственного многоканального сигнала. Бинауральная цепь генерирует бинауральный выходной сигнал путем обработки аудиосигнала. Обработка представляет бинауральную функцию передачи, обеспечивающую позицию виртуального звукового источника для аудиосигнала. Цепь измерения генерирует данные измерения, указывающие на характеристику акустической среды, и цепь определения определяет параметр акустической среды на основе данных измерения. Параметр акустической среды может, как правило, являться параметром реверберации, таким как время реверберации. Цепь адаптации адаптирует бинауральную функцию передачи динамически на основе параметра акустической среды. К примеру, адаптация может модифицировать параметр реверберации для большего совпадения с характеристиками реверберации акустической среды. 2 н. и 12 з.п. ф-лы, 7 ил. РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 595 943 C2 (51) МПК H04S 3/00 (2006.01) G10L 19/008 (2013.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ (21)(22) Заявка: ИЗОБРЕТЕНИЯ К ПАТЕНТУ 2013136390/08, 03.01.2012 (24) Дата начала отсчета срока действия патента: 03.01.2012 Приоритет(ы): (30) Конвенционный приоритет: (43) Дата публикации заявки: 10.02.2015 Бюл. № 4 (73) Патентообладатель(и): КОНИНКЛЕЙКЕ ФИЛИПС ЭЛЕКТРОНИКС Н.В. (NL) R U 05.01.2011 EP 11150155.7 (72) Автор(ы): ОМЕН Арнольдус Вернер Йоханнес (NL), БРЕБАРТ Дирк Ерун (NL), КОППЕНС Ерун Герардус Хенрикус (NL), СХЕЙЕРС Эрик Госейнус Петрус (NL) (56) Список документов, цитированных в отчете о поиске: Binaural reverberation using a modified Jot reverberator with frequency-dependent interaural coherence matching, Fritz Menzer, Christof Faller, AES Convention 126; ...

method of protecting spring reverberators from low-frequency vibrations

Audio signal processing method and audio signal processing device

FIELD: computer technology. SUBSTANCE: invention relates to the field of computer technology for processing audio data. The expected result is achieved by receiving a line input audio signal from a sound source, controlling the volume of the line input audio signal and generating an early reflected sound control signal in order to reproduce the early reflected sound of a predetermined space, which is the purpose for reproducing the sound field, using a line input audio signal with the volume obtained as a result of control. EFFECT: increase in the accuracy of providing the required sound field for an electronic musical instrument. 20 cl, 21 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 765 334 C1 (51) МПК G10L 19/008 (2013.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (52) СПК G10L 19/008 (2021.08) (21)(22) Заявка: 2021104066, 18.02.2021 (24) Дата начала отсчета срока действия патента: (73) Патентообладатель(и): ЯМАХА КОРПОРЕЙШН (JP) Дата регистрации: 28.01.2022 19.02.2020 JP 2020-025818 (45) Опубликовано: 28.01.2022 Бюл. № 4 2 7 6 5 3 3 4 R U (54) СПОСОБ ОБРАБОТКИ ЗВУКОВОГО СИГНАЛА И УСТРОЙСТВО ОБРАБОТКИ ЗВУКОВОГО СИГНАЛА (57) Реферат: Изобретение относится к области и формирования сигнала управления ранним вычислительной техники для обработки отраженным звуком, чтобы воспроизводить аудиоданных. Технический результат заключается ранний отраженный звук предварительно в повышении точности обеспечения требуемого определенного пространства, которое является звукового поля для электронного музыкального целью для воспроизведения звукового поля, с инструмента. Технический результат достигается использованием звукового сигнала линейного за счет приема звукового сигнала линейного входа с громкостью, полученной в результате входа от источника звука, управления управления. 2 н. и 18 з.п. ф-лы, 21 ил. громкостью звукового сигнала линейного входа Стр.: 1 C 1 C 1 Адрес для переписки: 129090, Москва, ул. Б. Спасская, 25, стр. 3, ООО " ...

Reverberator and method of reverberating audio signal

FIELD: physics, acoustics. SUBSTANCE: group of inventions relates to acoustics, particularly a reverberator and a method of reverberating an audio signal. The audio signal reverberator has a processor, which includes feedback loops with delay elements of two different frequency subband signals. The processor provides an audio signal with different delay loops to obtain reverberated frequency subband signals. The processor includes, for a first frequency subband signal, at least two frequency subband signals of a first delay line, having a first set of delay line taps which provide signals with delay by different delay taps, a first feedback loop connected to the delay line, and a first coupling unit for coupling signal outputs with the set of delay line taps. The processor also includes feedback loops with delay elements for a second frequency subband signal, at least two frequency subband signal of a second delay line, having a second set of delay line taps, which provide signals with delay by different delay taps, a second feedback loop connected to the second delay line and a second coupling unit for coupling signal outputs with the second set of delay line taps. EFFECT: high sound quality. 18 cl, 23 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 558 004 C2 (51) МПК G10K 15/12 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ (21)(22) Заявка: ИЗОБРЕТЕНИЯ К ПАТЕНТУ 2012118785/28, 06.10.2010 (24) Дата начала отсчета срока действия патента: 06.10.2010 Приоритет(ы): (30) Конвенционный приоритет: (72) Автор(ы): ВИЛКАМО Йуха (DE), НЕУГЕБАУЕР Бернхард (DE), ПЛОГШТИЕС Ян (DE) 21.10.2009 US 61/253,655 (43) Дата публикации заявки: 10.11.2013 Бюл. № 31 R U (73) Патентообладатель(и): Фраунхофер-Гезелльшафт цур Фёрдерунг дер ангевандтен Форшунг Е.Ф. (DE) (45) Опубликовано: 27.07.2015 Бюл. № 21 (86) Заявка PCT: C 2 C 2 (85) Дата начала рассмотрения заявки PCT на национальной фазе: 04.05.2012 EP 2010/064909 (06.10.2010) (87) Публикация заявки PCT: 2 5 5 ...

Reverberator and method for reverberating an audio signal

오디오 신호(5)를 반향하기 위한 반향기(10)는 반향된 주파수 서브밴드 신호들(27)을 획득하기 위해 다른 루프 지연들(23)에 의해 오디오 신호(5)를 나타내는 적어도 두 개의 다른 주파수 서브밴드 신호들(17)을 지연하기 위한 피드백 지연 루프 프로세서(20)를 포함한다. A reflector 10 for echoing an audio signal 5 includes at least two different frequencies representing the audio signal 5 by other loop delays 23 to obtain echoed frequency subband signals 27 And a feedback delay loop processor 20 for delaying the subband signals 17.

Fabricating method of thin film transistor

A method for manufacturing a TFT is provided to obtain large grains from a semiconductor layer, to improve the mobility of electrons and to reduce the leakage current by using a capping layer with an aiming refractive index. An amorphous silicon layer is formed on a substrate(101). A capping layer(104) is formed on the amorphous silicon layer. The capping layer has a refractive index range of 1.90 to 1.94. A metal catalytic layer(105) is formed on the capping layer. A heat treatment process is performed on the resultant structure to transform the amorphous silicon layer into a polycrystalline silicon layer(109).

Acoustic reproducing apparatus and method for enhancing low-frequency content

An acoustic enhancing apparatus and a method for enhancing low-frequency content are provided to prevent localization phenomenon in a head by giving externalization to an acoustic signal with enhanced low-frequency content. First and second gain control parts(110,120) control gain of first and second input signals respectively. A first addition part(130) provides a signal obtained by adding signals outputted from the first and second gain control part as an input signal of a low-frequency enhancing part(200). The low-frequency enhancing part delays and controls the gain of the signal inputted from the first addition part, and outputs a low-frequency enhanced signal by adding reflected sound signals for enhancing low-frequency contents. An externalization generation part(300) generates first and second output signals with externalization, by delaying the low-frequency enhanced signal by a first time and a second time.

Sound signal processing method and sound signal processing device

本发明实现针对电子乐器等的期望的声场的辅助。音信号处理方法对音信号进行线输入，对所述线输入的音信号进行音量控制，根据所述音量控制的音信号而生成初始反射音模拟信号。

Generating binaural audio in response to multichannel audio using at least one feedback delay network

Generating binaural audio in response to multi-channel audio using at least one feedback delay network

A method and system for generating a binaural signal in response to a set of channels of a multi-channel audio input signal comprises: applying a binaural room impulse response, BRIR, to each channel of the set, thereby generating filtered signals; and combining the filtered signals to generate the binaural signal, wherein applying the BRIR to each channel of the set comprises using a late reverberation generator (200) to apply, in response to control values asserted to the late reverberation generator (200), a common late reverberation to a downmix of the channels of the set, wherein the common late reverberation emulates collective macro attributes of late reverberation portions of single-channel BRIRs shared across at least some channels of the set, and wherein left-side channels of the multi-channel audio input signal are panned to a left channel of the downmix, and wherein right-side channels of the multi-channel audio input signal are panned to a right channel of the downmix.

Generation of binaural sound signal (brir) in response to multi-channel audio signal with use of feedback delay network (fdn)

FIELD: radio engineering, communication. SUBSTANCE: in the context of the method the input channels are treated in the first data processing channel to apply to each channel a portion of the direct feature and early reflections of a single channel BRIR for that channel, and the combined signal of these channels is processed in the second data channel containing at least one FDN, which uses a common late reverberation. Late reverberation emulates the collective macroscopic defining features of late reverberation portions of at least some of the single-channel BRIR features. EFFECT: ability to virtualize the generation of a binaural signal in response to channels of a multi-channel input audio signal applying the binaural impulse response of the room to each channel, which includes using at least one feedback delay network for applying a common late reverberation to the combined channel signal. 44 cl, 17 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 637 990 C1 (51) МПК H04S 3/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (21)(22) Заявка: 2016126479, 18.12.2014 (24) Дата начала отсчета срока действия патента: 18.12.2014 Дата регистрации: Приоритет(ы): (30) Конвенционный приоритет: (73) Патентообладатель(и): ДОЛБИ ЛАБОРАТОРИС ЛАЙСЭНЗИН КОРПОРЕЙШН (US) (56) Список документов, цитированных в отчете о поиске: WO 2012/093352 A1, 12.07.2012. WO (45) Опубликовано: 08.12.2017 Бюл. № 34 99/14983 A1, 25.03.1999. US 5,371,799 A1, 06.12.1994. RU 2011 105 972 A, 27.08.2012. (85) Дата начала рассмотрения заявки PCT на национальной фазе: 01.07.2016 (86) Заявка PCT: US 2014/071100 (18.12.2014) 2 6 3 7 9 9 0 03.01.2014 US 61/923,579; 29.04.2014 CN 201410178258.0; 05.05.2014 US 61/988,617 R U 08.12.2017 (72) Автор(ы): ЕНЬ Куань-Чиэф (US), БРЕБАРТ Дирк Дж. (AU), ДЭВИДСОН Грант А. (US), УИЛСОН Ронда (US), КУПЕР Дэвид М. (AU), ШУАН Чживэй (CN) (87) Публикация заявки PCT: 2 6 3 7 9 9 0 R U Адрес для переписки: 123242, Москва, Кудринская ...

An audio system and method of operation therefor

一种音频系统包括接收器（301），该接收器用于接收音频信号，例如音频对象或者空间多声道信号的某声道的信号。双耳电路（303）通过处理音频信号生成双耳输出信号。该处理代表提供用于音频信号的虚拟声源位置的双耳传递函数。测量电路（307）生成指示声学环境的特性的测量数据，并且确定电路（311）响应于测量数据而确定声学环境参数。声学环境参数典型地可以是回响参数，例如回响时间。适应电路（313）响应于声学环境参数而适应性调节双耳传递函数。例如，该适应性调节可以修改回响参数以便更密切地与声学环境的回响特性相似。

Generating binaural audio by using at least one feedback delay network in response to multi-channel audio

本公开涉及响应于多通道音频通过使用至少一个反馈延迟网络产生双耳音频。在一些实施例中，提供用于响应多通道音频信号的通道产生双耳信号的虚拟化方法，这些虚拟化方法向各通道应用双耳房间脉冲响应(BRIR)，包含通过使用至少一个反馈延迟网络(FDN)以向通道的下混应用公共晚期混响。在一些实施例中，输入信号通道在第一处理路径中被处理，以向各通道应用的用于该通道的单通道BRIR的直接响应和早期反射部分，并且，通道的下混在包含至少一个应用公共晚期混响的FDN的第二处理路径中被处理。典型地，公共晚期混响模仿单通道BRIR中的至少一些的晚期混响部分的共同宏观属性。其它方面是被配置为执行方法的任何实施例的耳机虚拟化器。

Sound reproduction apparatus and method of enhancing low frequency component

一种使用反射声音增强再现的声音信号中的低频分量的声音再现设备和方法。所述声音再现设备对输入信号进行延迟并控制其增益，产生预定数量的反射声音信号以增强低频分量，并输出具有增强低频分量的反射声音信号的和。

Generating a binaural audio signal in response to a multichannel audio signal using at least one feedback delay circuit

FIELD: acoustics. SUBSTANCE: invention relates to acoustics. A method for generating a binaural signal in response to a series of channels of a multi-channel audio input signal is to apply the binaural impulse response BRIR of the facility to each channel of the specified series, generating filtered signals as a result; and combining the filtered signals to generate a binaural signal. The BRIR application to each channel of the specified series involves the use of a late reverb generator with the application, in response to the control values entered in the late reverb generator, of the total late reverb to the reduced signal of the channels of the specified series, while the total late reverb emulates the collective macroscopic defining features of the late reverb parts of single-channel BRIRs shared on at least some channels of the series, and the reduced signal is the reduced stereo signal of the channels of the specified series. EFFECT: technical result consists in ensuring effective externalization. 14 cl, 17 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 747 713 C2 (51) МПК H04S 3/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (52) СПК H04S 3/004 (2021.02) (21)(22) Заявка: 2017138558, 18.12.2014 (24) Дата начала отсчета срока действия патента: Дата регистрации: 13.05.2021 03.01.2014 US 61/923,579; 29.04.2014 CN 201410178258.0; 05.05.2014 US 61/988,617 Номер и дата приоритета первоначальной заявки, из которой данная заявка выделена: 2016126479 03.01.2014 (43) Дата публикации заявки: 11.02.2019 Бюл. № 5 2 7 4 7 7 1 3 R U Адрес для переписки: 123242, Москва, пл. Кудринская, 1, а/я 35, "Михайлюк, Сороколат и партнеры патентные поверенные" (56) Список документов, цитированных в отчете о поиске: WO 2012093352 A1, 12.07.2012. F. Menzer, "Binaural Signal Processing Using Interaural Coherence Matching," Thèse No. 4643 (2010), Ecole Polytechnique Federale de Lausanne, April 2010. EP 1072089 B1, 09.03.2011. EP ...

Sound effect apparatus

내용 없음. No content.

Method for Processing an Audio Signal, Signal Processing Unit, Binaural Renderer, Audio Encoder and Audio Decoder

룸 임펄스 응답에 따라 오디오 신호(504) 처리하는 방법이 기술되었다. 오디오 신호(504)는 룸 임펄스 응답의 후기 반향과 별도로 룸 임펄스 응답의 초기 부분과 처리(502)되며, 후기 반향의 처리(514)는 스케일링된 반향 신호를 생성하는 것을 포함하며, 스케일링(526)은 오디오 신호(504)에 종속적이다. 오디오 신호(504)의 처리된 초기 부분(506)과 스케일링 된 반향 신호는 조합된다. A method of processing an audio signal 504 in response to a room impulse response has been described. The audio signal 504 is processed 502 with an initial portion of the room impulse response separately from the late echo of the room impulse response and the post echo processing 514 includes generating a scaled echo signal, 0.0 > 504 < / RTI > The processed initial portion 506 of the audio signal 504 and the scaled echo signal are combined.

Method for audio processing in accordance with impulse room characteristics, signal processing unit, audiocoder, audiodecoder and binaural rendering device

FIELD: physics. SUBSTANCE: audio signal is processed separately from the primary part and the late reverberation in the impulse response of the room, and the processed primary part of the audio signal and the reverbed signal are combined. The transition from the primary part to the late reverberation in the impulse response of the room is achieved when the correlation measure reaches a threshold value, the threshold value being set by the dependent correlation measure for the selected reflection from the primary reflections in the primary part of the impulse room characteristic. EFFECT: method solves the problem of providing the final audio signal provided to the user, the spatial sensation and the sound depth. 21 cl, 14 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 643 867 C2 (51) МПК G10K 15/12 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (52) СПК G10K 15/12 (2006.01) (21)(22) Заявка: 2016105519, 16.07.2014 (24) Дата начала отсчета срока действия патента: (73) Патентообладатель(и): ФРАУНХОФЕР-ГЕЗЕЛЛЬШАФТ ЦУР ФЕРДЕРУНГ ДЕР АНГЕВАНДТЕН ФОРШУНГ Е.Ф. (DE) Дата регистрации: 06.02.2018 (56) Список документов, цитированных в отчете о поиске: US 2008/0175396 A1, 24.07.2008. WO 22.07.2013 EP 13177362.4; 18.10.2013 EP 13189230.9 (43) Дата публикации заявки: 25.08.2017 Бюл. № 24 99/49574 A1, 30.09.1999. EP 2028884 A1, 25.02.2009. RU 2403674 C2, 10.11.2010. (45) Опубликовано: 06.02.2018 Бюл. № 4 (86) Заявка PCT: C 2 C 2 (85) Дата начала рассмотрения заявки PCT на национальной фазе: 22.02.2016 EP 2014/065227 (16.07.2014) (87) Публикация заявки PCT: 2 6 4 3 8 6 7 WO 2015/010983 (29.01.2015) R U 2 6 4 3 8 6 7 Приоритет(ы): (30) Конвенционный приоритет: R U 16.07.2014 (72) Автор(ы): ФЮГ Зимоне (DE), ПЛОГСТИС Ян (DE) Адрес для переписки: 129090, Москва, ул. Б.Спасская, 25, строение 3, ООО "Юридическая фирма Городисский и Партнеры" (54) СПОСОБ ДЛЯ ОБРАБОТКИ АУДИОСИГНАЛА В СООТВЕТСТВИИ С ИМПУЛЬСНОЙ ХАРАКТЕРИСТИКОЙ ПОМЕЩЕНИЯ, БЛОК ...

Sound effect compensation method and device, readable storage medium and terminal

本申请涉及一种音效补偿方法和装置、计算机可读存储介质、终端。该方法包括：根据音频处理装置当前输出的音频信号确定音频处理装置在用户耳道内的声学传输函数；根据声学传输函数获取当前音频处理装置的监测频响曲线；对比标准频响曲线与频响曲线以补偿音频信号，可以通过音频处理装置当前输出的音频信号获取用户的耳纹信息以及与耳纹信息相对应的监测频响曲线，根据标准频响曲线自动对监测频响曲线进行调整，以补偿音频信号的音色，从而适应于具有不同耳纹信息的用户，为不同的用户提供更好的听觉体验。

DEVICE FOR SIGNAL PROCESSING, SIGNAL PROCESSING METHOD AND PROGRAM

РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2020 112 255 A (51) МПК G10K 15/12 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ЗАЯВКА НА ИЗОБРЕТЕНИЕ (21)(22) Заявка: 2020112255, 05.10.2018 (71) Заявитель(и): СОНИ КОРПОРЕЙШН (JP) Приоритет(ы): (30) Конвенционный приоритет: 20.10.2017 JP 2017-203876 (85) Дата начала рассмотрения заявки PCT на национальной фазе: 26.03.2020 R U (43) Дата публикации заявки: 27.09.2021 Бюл. № 27 (72) Автор(ы): ЦУДЗИ Минору (JP), ТИНЕН Тору (JP), ФУКУЙ Такао (JP), ХАТАНАКА Мицуюки (JP) (86) Заявка PCT: (87) Публикация заявки PCT: WO 2019/078034 (25.04.2019) R U (54) УСТРОЙСТВО ДЛЯ ОБРАБОТКИ СИГНАЛА, СПОСОБ ОБРАБОТКИ СИГНАЛА И ПРОГРАММА (57) Формула изобретения 1. Устройство для обработки сигнала, содержащее модуль реверберационной обработки, который генерирует сигнал реверберационной составляющей на основе данных аудиообъекта и параметра реверберации для этого аудиообъекта. 2. Устройство для обработки сигнала по п. 1, дополнительно содержащее процессорный модуль отображения, который осуществляет процедуру отображения для сигнала реверберационной составляющей на основе параметра реверберации. 3. Устройство для обработки сигнала по п. 2, отличающееся тем, что параметр реверберации содержит информацию о местонахождении, указывающую позицию локализации звукового изображения реверберационной составляющей, и процессорный модуль отображения осуществляет процедуру отображения на основе информации о местонахождении. 4. Устройство для обработки сигнала по п. 3, отличающееся тем, что информация о местонахождении содержит информацию, указывающую абсолютную позицию локализации звукового изображения реверберационной составляющей. 5. Устройство для обработки сигнала по п. 3, отличающееся тем, что информацию о местонахождении содержит информацию, указывающую относительную позицию локализации, относительно аудиообъекта, звукового изображения реверберационной составляющей. 6. Устройство для обработки сигнала по п. 1, отличающееся тем, что ...

Sound effect apparatus

내용 없음. No content.

Resonance signal generator, electronic music device, resonance signal generation method and program

Audio signal processing method and audio signal processing device

Изобретение относится к способу обработки звукового сигнала и устройству обработки звукового сигнала. Способ обработки звукового сигнала содержит этапы, на которых: получают звуковой сигнал; получают данные импульсной характеристики, которые были измерены в предварительно определенном пространстве до получения звукового сигнала; формируют сигнал управления ранним отраженным звуком путем свертки данных импульсной характеристики раннего отраженного звука среди полученных данных импульсной характеристики в полученный звуковой сигнал; формируют сигнал управления реверберацией, не включающий в себя прямой звук, путем свертки данных импульсной характеристики реверберирующего звука среди полученных данных импульсной характеристики в полученный звуковой сигнал; выполняют первую обработку сигнала над сигналом управления ранним отраженным звуком; выполняют вторую обработку сигнала, отличную от первой обработки сигнала, над сигналом управления реверберацией; выводят сигнал управления реверберацией, прошедший вторую обработку сигнала, на по меньшей мере один первый громкоговоритель; и выводят сигнал управления ранним отраженным звуком, прошедший первую обработку сигнала, только на по меньшей мере один второй громкоговоритель. Технический результат – обеспечение лучшего пространственного расширения звука и качества звукового образа. 2 н. и 18 з.п. ф-лы, 19 ил. РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 762 879 C1 (51) МПК H04R 23/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (52) СПК H04R 23/00 (2021.08) (21)(22) Заявка: 2021104064, 18.02.2021 (24) Дата начала отсчета срока действия патента: (73) Патентообладатель(и): ЯМАХА КОРПОРЕЙШН (JP) Дата регистрации: 23.12.2021 19.02.2020 JP 2020-025816 (45) Опубликовано: 23.12.2021 Бюл. № 36 2 7 6 2 8 7 9 R U (54) Способ обработки звукового сигнала и устройство обработки звукового сигнала (57) Реферат: Изобретение относится к способу обработки в полученный звуковой сигнал; выполняют ...

Device and method for generation of output signals based on audio source signal, audio reproductive system and loudspeaker signal

FIELD: physics.SUBSTANCE: invention relates to audio signal processing means. At least one signal of the audio source and the reverberated audio signal are delayed and combined with the delay line to obtain a combined signal. Filtering the combined signal from the delay line to obtain an output signal, wherein the first plurality of output signals is obtained from the second plurality of delay paths, wherein each delay path has a delay line. Reverberate the first plurality of output signals to obtain a third plurality of reverberated audio signals. Combined signal comprises a signal portion of the audio source and a portion of the reverberant signal. Delay line comprises a sixth plurality of input taps which are configured to receive the audio source signal or its weighted version. Audio source signal or its weighted version is transmitted to one of the sixth plurality of input taps and based on the first position of the virtual audio source in the virtual playback room and no signal of the audio source or its weighted version is transmitted to the other input tap of the sixth plurality of input taps.EFFECT: technical result consists in improvement of efficiency of obtaining reverberated signals.21 cl, 14 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 686 026 C2 (51) МПК H04S 7/00 (2006.01) G10K 15/12 (2006.01) H04S 3/02 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (52) СПК H04S 3/02 (2018.08); H04S 7/00 (2018.08); G10K 15/12 (2018.08) (21) (22) Заявка: 2017119648, 29.10.2015 (24) Дата начала отсчета срока действия патента: 29.10.2015 23.04.2019 07.11.2014 EP 14192213.8 (43) Дата публикации заявки: 07.12.2018 Бюл. № 34 (73) Патентообладатель(и): ФРАУНХОФЕР-ГЕЗЕЛЛЬШАФТ ЦУР ФЕРДЕРУНГ ДЕР АНГЕВАНДТЕН ФОРШУНГ Е.Ф. (DE) (56) Список документов, цитированных в отчете о поиске: US 5491754 A1, 13.02.1996. US 2009/ (45) Опубликовано: 23.04.2019 Бюл. № 12 (86) Заявка PCT: EP 2015/075141 (29.10.2015) R U 2 6 8 6 0 2 6 C 2 (87) ...

Audio system and operational approach thereof

一种音频系统包括接收器（301），该接收器用于接收音频信号，例如音频对象或者空间多声道信号的某声道的信号。双耳电路（303）通过处理音频信号生成双耳输出信号。该处理代表提供用于音频信号的虚拟声源位置的双耳传递函数。测量电路（307）生成指示声学环境的特性的测量数据，并且确定电路（311）响应于测量数据而确定声学环境参数。声学环境参数典型地可以是回响参数，例如回响时间。适应电路（313）响应于声学环境参数而动态地适应性调节双耳传递函数。例如，该适应性调节可以修改回响参数以便更密切地与声学环境的回响特性相似。

Method for processing an audio signal, signal processing unit, binaural renderer, audio encoder and audio decoder

Device and method of determining indicator for perceptible reverberation level, audio processor and signal processing method

FIELD: physics. SUBSTANCE: device for determining an indicator of a perceptible reverberation level in a composite signal, consisting of a forward signal component and a reverberation signal component, comprises a volume model processor, which comprises a perceptional filter stage for filtering the "dry" signal component, the reverberation signal component or the composite signal, wherein the perceptional filter stage is configured to simulate the audio perception mechanism of the object to obtain a filtered forward signal, a filtered reverberation signal or a filtered composite signal, a volume estimating unit for estimating a first volume indicator using the filtered forward signal and for estimating a second volume indicator using the filtered reverberation signal or the filtered composite signal, where the filtered composite signal is obtained by superposition of the forward signal component and the reverberation signal component, an integration unit for integrating the first and second volume indicators. EFFECT: improved characteristics of perceptible sound. 16 cl, 17 dwg, 3 tbl РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (51) МПК H04S 5/00 (13) 2 550 528 C2 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ (21)(22) Заявка: ИЗОБРЕТЕНИЯ К ПАТЕНТУ 2013144058/08, 24.02.2012 (24) Дата начала отсчета срока действия патента: 24.02.2012 Приоритет(ы): (30) Конвенционный приоритет: (43) Дата публикации заявки: 10.04.2015 Бюл. № 10 (56) Список документов, цитированных в отчете о поиске: WO 2009/039897 A1, 02.04.2009. RU 2330390 C2, 27.07.2008. EP2154911 A1, 17.02.2010. EP 1565036 A2, 17.08.2005 (85) Дата начала рассмотрения заявки PCT на национальной фазе: 02.10.2013 2 5 5 0 5 2 8 (45) Опубликовано: 10.05.2015 Бюл. № 13 (73) Патентообладатель(и): ФРАУНХОФЕР-ГЕЗЕЛЛЬШАФТ ЦУР ФЕРДЕРУНГ ДЕР АНГЕВАНДТЕН ФОРШУНГ Е.Ф. (DE) R U 02.03.2011 US 61/448,444; 27.06.2011 EP 11171488.7 (72) Автор(ы): УЛЕ Кристиан (DE), ПАУЛУС Йоуни (DE), ХЕРРЕ Юрген (DE), ...

Sound signal processing method and sound signal processing device

【課題】初期反射音の音質を向上する音信号処理方法および音信号処理装置を提供する。【解決手段】音信号処理装置１０には、複数の音源ＯＢＪ１－ＯＢＪ９６の音信号Ｓ１－Ｓ９６が入力される。グルーピング部４０は、音源ＯＢＪ１－ＯＢＪ９６を、複数の領域Ａｒｅａ１－Ａｒｅａ８にグルーピングし、音源ＯＢＪ１－ＯＢＪ９６の音信号Ｓ１－Ｓ９６を用いて、領域Ａｒｅａ１－Ａｒｅａ８毎の領域別音信号ＳＡ１－ＳＡ８を生成する。初期反射音制御信号生成部５０は、再生空間に配置されたスピーカＳＰ１－ＳＰ６４の位置と仮想空間の幾何学形状とを用いてグループ毎に虚音源を設定し、仮想空間での初期反射音を模擬する初期反射音制御信号ＥＲ１－ＥＲ６４を生成する。この際、初期反射音制御信号生成部は、初期反射音制御信号に対して、所望の音色調整を行う。【選択図】図１

Method for audio signal processing and audio signal processing apparatus

FIELD: computing technology.SUBSTANCE: invention relates to the field of computing technology, namely, to means of audio signal processing. The method for audio signal processing includes the stages of receiving multiple audio signals collected by multiple microphones placed in space, respectively, adjusting the corresponding levels of the multiple audio signals in accordance with the corresponding positions of the multiple microphones, mixing the multiple audio signals with the adjusted corresponding levels, wherein the stage of adjusting the corresponding levels of the multiple audio signals includes the stage of adjusting each of the corresponding levels of the multiple audio signals in accordance with the distance from each of the corresponding positions of the multiple microphones to a loudspeaker outputting the reflected sound.EFFECT: increase in the accuracy of processing reflected signals.12 cl, 21 dwg РОССИЙСКАЯ ФЕДЕРАЦИЯ (19) RU (11) (13) 2 770 438 C1 (51) МПК H04R 5/04 (2006.01) H04S 5/02 (2006.01) H04S 7/00 (2006.01) G06K 1/00 (2006.01) ФЕДЕРАЛЬНАЯ СЛУЖБА ПО ИНТЕЛЛЕКТУАЛЬНОЙ СОБСТВЕННОСТИ (12) ОПИСАНИЕ ИЗОБРЕТЕНИЯ К ПАТЕНТУ (52) СПК H04R 5/04 (2022.02); H04S 5/02 (2022.02); H04S 7/00 (2022.02); G06K 1/00 (2022.02) (21)(22) Заявка: 2021104065, 18.02.2021 (24) Дата начала отсчета срока действия патента: (73) Патентообладатель(и): ЯМАХА КОРПОРЕЙШН (JP) Дата регистрации: 18.04.2022 19.02.2020 JP 2020-025817 (45) Опубликовано: 18.04.2022 Бюл. № 11 2 7 7 0 4 3 8 R U (54) СПОСОБ ОБРАБОТКИ ЗВУКОВОГО СИГНАЛА И УСТРОЙСТВО ОБРАБОТКИ ЗВУКОВОГО СИГНАЛА (57) Реферат: Изобретение относится к области микрофонов, микшируют множество звуковых вычислительной техники, а именно к средствам сигналов с отрегулированными обработки звуковых сигналов. Технический соответствующими уровнями, причем этап, на результат заключается в повышении точности котором регулируют соответствующие уровни обработки отраженных сигналов. Способ множества звуковых сигналов, включает в себя обработки ...

Signal processing equipment and methods, and programs

本技術は、距離感制御をより効果的に実現することができるようにする信号処理装置および方法、並びにプログラムに関する。信号処理装置は、オーディオオブジェクトのオブジェクトオーディオデータと、オーディオオブジェクトについてのリバーブパラメタとに基づいてリバーブ成分の信号を生成するリバーブ処理部を備える。本技術は信号処理装置に適用することができる。 The present technology relates to signal processing devices and methods, and programs that enable more effective distance control. The signal processing device includes a reverb processing unit that generates a signal of a reverb component based on the object audio data of the audio object and the reverb parameter for the audio object. This technology can be applied to signal processing equipment.

Electroacoustic reverberation support device

High sound quality improving device and method, and computer program

High sound quality improving device and method, and computer program

高音質化装置は、元の音声信号に対して所定種類の処理が施された処理済音声信号を、所定時間遅延させることで基本音信号を生成し、処理済音声信号を所定時間より短い時間だけ遅延させることで後方音信号を生成し、処理済音声信号を所定時間より長い時間だけ遅延させることで前方音信号を生成するディレイ手段を備える。そして、少なくとも生成された後方音信号及び生成された前方音信号に基いて、基本音信号が示す基本音に対する効果音を示す効果音信号を生成する効果音生成手段と、生成された効果音信号を基本音信号に合成する合成手段とを備える。 The sound quality improvement device generates a basic sound signal by delaying a processed sound signal obtained by performing a predetermined type of processing on the original sound signal for a predetermined time, and the processed sound signal is shorter than the predetermined time. Delay means for generating a backward sound signal by delaying the processed sound signal by delaying the processed sound signal by a time longer than a predetermined time. Then, based on at least the generated rear sound signal and the generated front sound signal, a sound effect generating means for generating a sound effect signal indicating a sound effect for the basic sound indicated by the basic sound signal, and the generated sound effect signal And a synthesizing means for synthesizing the sound into a basic sound signal.

Audio signal processing system and method

A method, an apparatus, and a software product to process a plurality of input audio signals (15-19). The apparatus accepts a plurality of input signals (15-19) and includes a multi-input, multi-output reverberator (14) arranged to, generate a set of output signals (35-39) including delayed reverberation components simulating the reverberations a listener is likely to hear in a listening environment. The apparatus further includes a multi-input, two-output filter (20-24) accepting the outputs of the reverberator and the plurality of input terminals, providing outputs for the left and right ears, and configured to implement a set of head related transfer functions corresponding to a listening environment and a set of directions of a listener in the listening environment. The apparatus is such that a listener listening to the outputs through headphones (47-48) has the sensation of listening to the plurality of input audio signals as if they are emanating from a plurality of loudspeakers spatially located in the listening environment at a corresponding plurality of directions.

Patent FR2500194B1

Improvements to acoustic systems

Improvements to electroacoustic systems

Process for improving the sound in t. s. f.

Sound reproducing device for studying room acoustics

Device for the artificial reverberation of sounds

Device for adding echo to a signal

Generating binaural audio in response to multi-channel audio using at least one feedback delay network

【課題】音響環境のよりよいマッチングおよびより自然に聞こえる出力を可能にするマルチチャネル・オーディオに応答したバイノーラル生成方法を提供する。【解決手段】バイノーラル信号を生成する仮想化方法は、各チャネルにバイノーラル室内インパルス応答（ＢＲＩＲ）を適用する。これは、チャネルのダウンミックスに共通の後期残響を加えるよう少なくとも一つのフィードバック遅延ネットワーク（ＦＤＮ）を使う。入力信号チャネルは、第一の処理経路において処理されて、各チャネルに、そのチャネルについての単一チャネルＢＲＩＲの直接応答及び早期反射部分を適用する。チャネルのダウンミックスは、共通の後期残響を加える一つのＦＤＮを含む第二の処理経路において処理される。共通の後期残響は、単一チャネルＢＲＩＲのうちの少なくともいくつかのＢＲＩＲの後期残響部分の集団的なマクロ属性をエミュレートする。【選択図】図１０

Reverberation apparatus

ABSTRACT A reverberation apparatus is formed of bucket brigade delay circuit having its input connected to a signal source. An output of the delay circuit is connected through a spring-type reverberator to one of two inputs of an adder, the other being connected to the signal input. The output of the adder provides the desired reverberation sound. In one embodiment, two series connected delay circuits and reverberators are provided, each connecting to a respective input of a respective adder. The other inputs of each adder are connected to the signal source and the outputs of the respective adders provide separate stereophonic-like left and right channel outputs.

ARTIFICIAL REVERBERATION DEVICE

DISPOSITIF SERVANT A PRODUIRE UN EFFET DE REVERBERATION ARTIFICIELLE ET COMPORTANT UNE LIGNE DE RETARDEMENT RETROCOUPLEE. POUR OBTENIR PLUSIEURS DUREES D'ECHO, LADITE LIGNE EST COMPLETEE PAR UNE AUTRE LIGNE DE RETARDEMENT MUNIE DE BRANCHES DE REACTION, CETTE AUTRE LIGNE DONNANT LIEU A UNE DUREE D'ECHO EGALE A LA SOMME DE CELLES DE LA PREMIERE LIGNE ET DE L'AUTRE LIGNE DE RETARDEMENT. APPLICATION: EQUIPEMENTS DE SALLES DE SPECTACLES. DEVICE SERVING TO PRODUCE AN ARTIFICIAL REVERBERATION EFFECT AND INCLUDING A RETROCOUPLE DELAY LINE. TO OBTAIN SEVERAL ECHO DURATIONS, THE SAID LINE IS COMPLETED BY ANOTHER DELAY LINE EQUIPPED WITH REACTION BRANCHES, THIS OTHER LINE GIVING PLACE TO AN ECHO DURATION EQUAL TO THE SUM OF THOSE OF THE FIRST LINE AND THE OTHER DELAY LINE. APPLICATION: SHOWROOM EQUIPMENT.

BINURICULAR SOUND REPRODUCTION SYSTEM WITH ACOUSTIC REVERBERATION UNIT

Method for processing an audio signal in accordance with a room impulse response, signal processing unit, audio encoder, audio decorder, and binaural renderer

本發明揭露一種根據空間脈衝響應(434)處理一音源訊號(400)的方法。音源訊號(400)係與空間脈衝響應(434)之初期部分以及晚期迴響音分别地處理(422，424)，並結合(432)音源訊號之經處理的初期部分(428)以及迴響訊號(430)。當一相關性量測到達門檻值，則表示空間脈衝響應中從初期部分轉換成晚期迴響音，此門檻值之設定係取決於從空間脈衝響應之初期部分中選擇複數個初期反射音中的其中一個的相關性量測。

Method for the reverberation of sounds in a loudspeaker and loudspeaker for carrying out this method or similar method

Method for synthesizing auditorium sound

METHOD AND DEVICE FOR ACHIEVING AN ARTIFICIAL REVERBERATION

Reverberation loud-speaker assembly

Improvements in or relating to arrangements for use in producing artificial acousticreverberation

961,207. Reverberation systems. PHILIPS ELECTRICAL INDUSTRIES Ltd. Nov. 15, 1960 [Nov. 18, 1959], No. 39200/60. Heading H4R. In apparatus for use in producing artificial acoustic reverberation from audio-frequency electrical signals, an endless magnetic record carrier 10 (Fig. 1) co-operates with a recording head 1 and a plurality of reproducing heads 2, 3, 4, .5 each connected to recording head 1 through a respective feedback channel including an amplifier 9, 8, 7, 6 so that signals from the feedback channels are recorded on the carrier 10, and the gain of each feedback channel is automatically and periodically varied at a frequency below the audibility range, the gains of the different feedback channels being varied at frequencies which differ from each other. For automatically and periodically varying the gain of each feedback channel oscillators 16, 15, 14, 13 are provided which apply a sinusoidal bias potential to the first valve of each amplifier. Such first valve is of the variable-mu type. Frequencies of 5, 7, 9, 11 cycles per second are used for the four channels. Each reproducing head 2, 3, 4, 5 applies signals to a loud-speaker or group of loud-speakers after amplification. For simplicity only loud-speaker 12 and amplifier 11, coupled to head 5, are shown in Fig. 1. The varying bias potentials need not be sinusoidal. The reverberation effect may be controlled by control of the mean amplification factor of one or more of the feedback channels. Record carrier 10 may be a tape or a drum or any other rotating medium having magnetic material on its surface.

SOUND EFFECT DEVICE

Echo device

Filtering with binaural room impulse responses with content analysis and weighting

一种包括一或多个处理器的装置经配置以将适应性地确定的权数应用到音频信号的多个信道以产生所述音频信号的多个适应性地加权的信道。所述处理器经进一步配置以组合所述音频信号的所述多个适应性地加权的信道中的至少两个以产生组合信号。所述处理器经进一步配置以将立体声房间脉冲响应滤波器应用到所述组合信号以产生立体声音频信号。

SOUND SPOTLIGHT WITH ROOM EFFECT, OPTIMIZED IN COMPLEXITY

L'invention concerne une spatialisation sonore, avec l'application d'au moins une fonction de transfert à effet de salle à au moins un signal sonore. Cette application revient à multiplier, dans le domaine spectral, des composantes spectrales du signal sonore par les composantes spectrales d'un filtre correspondant à la fonction de transfert, chaque composante spectrale du filtre comportant une évolution temporelle dans une représentation temps-fréquence. En particulier, les composantes spectrales du filtre sont ignorées, pour les multiplications de composantes précitées, au-delà d'une fréquence seuil (Fcd(1), Fcg(1), Fcd(2), Fcg(2)) et après au moins un instant donné (m=1, m=2) dans cette représentation temps-fréquence. The invention relates to sound spatialization, with the application of at least one room effect transfer function to at least one sound signal. This application amounts to multiplying, in the spectral domain, the spectral components of the sound signal by the spectral components of a filter corresponding to the transfer function, each spectral component of the filter comprising a temporal evolution in a time-frequency representation. In particular, the spectral components of the filter are ignored, for the aforementioned component multiplications, beyond a threshold frequency (Fcd (1), Fcg (1), Fcd (2), Fcg (2)) and after minus a given moment (m = 1, m = 2) in this time-frequency representation.

Electro-acoustical system

A method and acoustical system for processing sound signals according to the principles of the acoustic holography of sound wave field extrapolation.

Magnetic recorder with echo effect

Electronic stereo reverberation device with doubler

ABSTRACT OF THE DISCLOSURE A stereo or two channel electronic reverberation device is disclosed comprising an analog delay device which receives audio signals and provides delayed output signals at a plurality of outputs, with the time delay period of each output being different from the delay period of other outputs. Two summing devices each receive inputs from different combinations of the analog delay device outputs to provide two different signals having different reverb com-ponents. An additional output delay device is also provided with receives the last output from the analog delay device ? delays this signal a time period substantially greater than the time period between any two adjacent analog delay device outputs, and provides this substantially greater delayed signal to only one of said summing devices. The circuit includes a synthetic doubler which provides an output cyclicly varying in pitch from its input. Two output mixers provide reverb alone, doubler signal alone or both reverb and doubling. A timed turn on gate at the input of the analog delay device substantially eliminates unwanted noise signals of short duration.

Filtering with binaural room impulse responses with content analysis and weighting

하나 이상의 프로세서들을 포함하는 디바이스는, 적응적으로 결정된 가중치들을 오디오 신호의 복수의 채널들에 적용하여 오디오 신호의 복수의 적응적으로 가중화된 채널들을 생성하도록 구성된다. 프로세서들은 또한 오디오 신호의 복수의 적응적으로 가중화된 채널들 중 적어도 2개를 결합하여 결합된 신호를 생성하도록 구성된다. 프로세서들은 또한 바이노럴 룸 임펄스 응답 필터를 결합된 신호에 적용하여 바이노럴 오디오 신호를 생성하도록 구성된다. A device including one or more processors is configured to apply adaptively determined weights to a plurality of channels of an audio signal to generate a plurality of adaptively weighted channels of the audio signal. The processors are also configured to combine at least two of the plurality of adaptively weighted channels of the audio signal to produce a combined signal. The processors are also configured to apply a binaural room impulse response filter to the combined signal to produce a binaural audio signal.

Method and apparatus for sound externalization in frequency domain

음장효과 생성 방법 및 장치가 개시된다. 음장효과 생성 장치는 직접 신호로부터 주파수 변환된 주파수 계수를 생성하고, 주파수 계수로부터 반사 신호를 생성하고, 주파수 계수 및 반사 신호를 이용하여 출력 신호를 생성하고, 출력 신호를 역 주파수 변환할 수 있다. A method and an apparatus for generating a sound field effect are disclosed. The sound field effect generating apparatus generates a frequency coefficient frequency-converted from the direct signal, generates a reflection signal from the frequency coefficient, generates an output signal using the frequency coefficient and the reflection signal, and reverse-frequency-converts the output signal. MDCT, 음장효과, 주파수 계수, 반사 신호 MDCT, sound field effect, frequency coefficient, reflection signal

Device for adding echoes to a sound signal