1167 results about "Reverberation" patented technology

A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

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

A neural network is provided for recognition and enhancement of multi-channel sound signals received by multiple microphones, which need not be aligned in a linear array in a given environment. Directions and distances of sound sources may also be detected by the neural network without the need for a beamformer connected to the microphones. The neural network may be trained by knowledge gained from free-field array impulse responses obtained in an anechoic chamber, array impulse responses that model simulated environments of different reverberation times, and array impulse responses obtained in actual environments.

An electronic circuit for adding the effects of reverberation to a high impedance signal produced from an external audio source such as a guitar pickup or a high impedance microphone. The electronic circuit comprises a reverberation effects circuit having pre-amplifier / driver and recovery amplifier sections and a spring reverberation device coupled thereinbetween and a reverberation effects bypass at the circuit's input, prior to switchably passing the audio signal into an input of the pre-amplifier / driver section which comprises a low impedance, high current output for input into the spring reverberation device having an output for passing a low impedance signal to the recovery amplifier section, which suitably serves in increasing the impedance of the signal to a predetermined level acceptable for input into an external sound device such as an amplifier having channel inputs and audio control capabilities, and a power supply circuit having means for switching between a dc voltage source and an ac voltage source and supplying ±9 volts to integrated circuits (ICs) included in the pre-amplifier / driver and recovery amplifier sections of the reverberation effects circuit.

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

Embodiments disclosed herein enable detection and improvement of the quality of the audio signal using a mobile device by determining the loss in the audio signal and enhancing audio by streaming the remainder portion of audio. Embodiments disclosed herein enable an improvement in the sound quality rendered by rendering devices by emitting an test audio signal from the source device, measuring the test audio signal using microphones, detecting variation in the frequency response, loudness and timing characteristics using impulse responses and correcting for them. Embodiments disclosed herein also compensate for the noise in the acoustic space by determining the reverberation and ambient noise levels and their frequency characteristics and changing the digital filters and volumes of the source signal to compensate for the varying noise levels.

An audio system comprises a receiver (301) for receiving an audio signal, such as an audio object or a signal of a channel of a spatial multi-channel signal. A binaural circuit (303) generates a binaural output signal by processing the audio signal. The processing is representative of a binaural transfer function providing a virtual sound source position for the audio signal. A measurement circuit (307) generating measurement data indicative of a characteristic of the acoustic environment and a determining circuit (311) determines an acoustic environment parameter in response to the measurement data. The acoustic environment parameter may typically be a reverberation parameter, such as a reverberation time. An adaptation circuit (313) adapts the binaural transfer function in response to the acoustic environment parameter. For example, the adaptation may modify a reverberation parameter to more closely resemble the reverberation characteristics of the acoustic environment.

Visual information is used to alter or set an operating parameter of an audio signal processor, other than a beamformer. A digital camera captures visual information about a scene that includes a human speaker and / or a listener. The visual information is analyzed to ascertain information about acoustics of a room. A distance between the speaker and a microphone may be estimated, and this distance estimate may be used to adjust an overall gain of the system. Distances among, and locations of, the speaker, the listener, the microphone, a loudspeaker and / or a sound-reflecting surface may be estimated. These estimates may be used to estimate reverberations within the room and adjust aggressiveness of an anti-reverberation filter, based on an estimated ratio of direct to indirect (reverberated) sound energy expected to reach the microphone. In addition, orientation of the speaker or the listener, relative to the microphone or the loudspeaker, can also be estimated, and this estimate may be used to adjust frequency-dependent filter weights to compensate for uneven frequency propagation of acoustic signals from a mouth, or to a human ear, about a human head.

A neural network classifier provides the ability to separate and categorize multiple arbitrary and previously unknown audio sources down-mixed to a single monophonic audio signal. This is accomplished by breaking the monophonic audio signal into baseline frames (possibly overlapping), windowing the frames, extracting a number of descriptive features in each frame, and employing a pre-trained nonlinear neural network as a classifier. Each neural network output manifests the presence of a pre-determined type of audio source in each baseline frame of the monophonic audio signal. The neural network classifier is well suited to address widely changing parameters of the signal and sources, time and frequency domain overlapping of the sources, and reverberation and occlusions in real-life signals. The classifier outputs can be used as a front-end to create multiple audio channels for a source separation algorithm (e.g., ICA) or as parameters in a post-processing algorithm (e.g. categorize music, track sources, generate audio indexes for the purposes of navigation, re-mixing, security and surveillance, telephone and wireless communications, and teleconferencing).

A signal processing system detects reverberation. The system may suppress the reverberation and improve signal quality. The system analyzes frequency bands of an input signal to determine whether reverberation characteristics are present. When reverberation is detected, the system may attenuate the reverberant frequency band to reduce or eliminate the reverberation.

Acoustic imaging provides images with reduced reverberation noise and images of nonlinear scattering and propagation parameters of the object. The received signal from transmitted dual frequency band acoustic pulse complexes is processed with overlapping high and low frequency pulses. The high frequency pulse is used for image reconstruction and the low frequency pulse manipulate the nonlinear scattering and / or propagation properties of the high frequency pulse. The scattered signal from a single dual band pulse complex for filtering in the fast time (depth time) domain provides a signal with suppression of reverberation noise with 1st harmonic sensitivity and increased spatial resolution. Through filtering in the pulse number coordinate and corrections for nonlinear propagation delays, a linear back scattering signal with suppressed pulse reverberation noise, a nonlinear back scattering signal, and quantitative nonlinear forward propagation and scattering parameters are extracted.

The present invention discloses a locating and tracking method based on a sound source array, comprising the following steps: S1, acquiring on-site sound through a quintuple microphone array, and pre-processing a sound signal acquired by each microphone in the quintuple microphone array to obtain an audio signal; S2, performing sound source locating for the audio signals according to arrival time delay of the audio signals between the microphones and positional information of the microphone array, so as to calculate a pitch angle, an azimuth angle and an object distance; and S3, moving and turning a locating and tracking apparatus to arrive to a sound source position. The locating and tracking method of the present invention performs related processing for influences of non-gauss noises, coherent noises and indoor reverberation of the sound source on accurate locating, thereby improving accuracy of locating of the sound source.

A scheme for stereo and multi-channel synthesis of inter-channel correlation (ICC) (normalized cross-correlation) cues for parametric stereo and multi-channel coding. The scheme synthesizes ICC cues such that they approximate those of the original. For that purpose, diffuse audio channels are generated and mixed with the transmitted combined (e.g., sum) signal(s). The diffuse audio channels are preferably generated using relatively long filters with exponentially decaying Gaussian impulse responses. Such impulse responses generate diffuse sound similar to late reverberation. An alternative implementation for reduced computational complexity is proposed, where inter-channel level difference (ICLD), inter-channel time difference (ICTD), and ICC synthesis are all carried out in the domain of a single short-time Fourier transform (STFT), including the filtering for diffuse sound generation.

An ultrasonic sensor includes a piezoelectric vibrator, a reverberation measuring device and a compensating device. The piezoelectric vibrator vibrates to transmit an ultrasonic wave, receives a reflected wave of the transmitted ultrasonic wave and produces an output signal corresponding to vibration of the piezoelectric vibrator. The reverberation measuring device measures a reverberation period of the piezoelectric vibrator from the output signal. The compensating device compensates for an operation characteristic of the piezoelectric vibrator in accordance with the measured reverberation period. Thus, the ultrasonic sensor can decrease a reverberation period, which may cause reduction in detection accuracy and impossibility of detection of an obstacle at a short distance.

The present invention is directed to a telephone equipped with multiple microphones that provides improved performance during operation of the telephone in a speaker-phone mode. For example, the multiple microphones can be used to improve voice activity detection, which in turn, can improve echo cancellation. In addition, the multiple microphones can be configured as an adaptive microphone array and used to reduce the effects of (i) room reverberation, when a near-end user is speaking, and / or (ii) acoustic echo, when a far-end user is speaking.

The invention discloses a karaoke in-pair singing method based on an internet and a system thereof. A vocal accompaniment user terminal plays instrument accompaniment of a song as well as reverberation and audio mixing processing is carried out on the instrument accompaniment of the song and a part sung by the vocal accompaniment user, so that an audio rack with the instrument accompaniment and the vocal accompaniment of the song is formed; the audio rack with the instrument accompaniment and the vocal accompaniment of the song is forwarded to a chorus user terminal through a transfer server; the chorus user terminal carries out decoding on the audio rack with the instrument accompaniment and the vocal accompaniment of the song, plays the audio track and carries out reverberation and audio mixing processing on the audio rack with the instrument accompaniment and the vocal accompaniment of the song and a part sung by the chorus user, thereby forming an audio rack with the instrument accompaniment, the vocal accompaniment and the chorus of the song; group sending is carried out on the audio rack with the instrument accompaniment, the vocal accompaniment and the chorus of the song to a listening client by a group sending server; and listening is carried out on the client as well as decoding is carried out on the audio rack with the instrument accompaniment, the vocal accompaniment and the chorus of the song to carry out playing. In addition, the invention also discloses a karaoke in-pair singing system based on the internet. According to the invention, an ideal in-pair singing effect can be provided for a user in a complicated environment with a large delay.

The invention provides a SRP-PHAT multi-source spatial positioning method. The method comprises the steps that the number and spatial positions of all microphones in a uniform circular microphone array are assumed to be unchanged in the data obtaining process at first, the isotropous microphones are evenly distributed on a circumference which has the radius r and is located on an x-y plane, the direction of arrival of a plane wave s is expressed by polar coordinates, the original point of the coordinate system is located on a circle center position of the circular array, multiple sound source signals are divided into non-overlapped time frequency point sets, each time frequency window contains only one movable source signal, and weak W orthogonal separation conditions are met; a Hamming window is selected, a controllable response power function is calculated and a target function is obtained through a SRP-PHAT algorithm, wave beams are controlled to carry out scanning in all the possible receiving directions, and the wave beams output the direction value with the maximum power to obtain the direction of a sound source, so that the DOA estimation of the multiple sound sources has the better separating performance in the strong noise and moderate reverberation acoustic environment, the real peak value is obviously outstanding, and high positioning precision is achieved.

An audio equivalent of a video zoom feature for video recording and communication applications, as well as video post production processes. The audio zoom may operate in conjunction with a video zoom feature or independently. The audio zoom may be achieved by controlling reverberation effects of a signal, controlling a gain of the signal, as well as controlling the width of a directional beam from which is used to select the particular audio component to focus on. The audio zoom may operate in response to user input, such as a user selection of a particular direction, or automatically based a current environment or other factors.

The invention provides a double-microphone speech enhancer, which comprises a double-microphone array module, a delay compensation module, a cepstrum field dereverberation module, a speech enhancement processing module and an output module, wherein the cepstrum field dereverberation module has the effect of dereverberation. The invention also provides a speech enhancement method based on the double-microphone speech enhancer. The invention adopts the cepstrum field dereverberation module to carry out beam formation and low-pass filtering, and can effectively eliminate the affection of room reverberation, and meanwhile, by using a speech enhancement algorithm, background noise can be further suppressed.

To provide a loudspeaker broadcasting system and a loudspeaker broadcasting apparatus which can perform a loudspeaker broadcasting of appropriate contents at each block without reducing sound clearness. Bi-directional speakers 1 which performs sound-reinforcement having high intelligibility, and fire detecting sensors 3 which detect fire occurrence in a tunnel 10a, 10b and send fire occurrence signals are placed in the tunnel 10a, 10b in the tunnel extending direction. In an emergency such as a fire, emergency broadcasting is performed through each bi-directional speaker 1 at each of the predetermined blocks 5 in the tunnel 10a, 10b. The emergency broadcasting is different in contents at each block, and it is alternately performed at the adjacent blocks 5. Further, the frequency components having a long reverberation time is removed from the loudspeaker broadcasting through the bi-directional speaker 1.

A leadless Deep Brain Stimulation system includes an implantable stimulator configured to convert incoming acoustic energy into an electric stimulation signal. The acoustic energy is provided by one or more external transmitters acoustically coupled to a head of the subject. To focus ultrasound energy onto the location of the stimulator, a wireless beacon (including a piezoelectric receiving transducer and an RF emitting antenna) is incorporated with the stimulator for providing an electromagnetic feedback signal to the external controller. The external controller is configured to send an initial unfocused acoustic signal towards the stimulator and receive the electromagnetic radiofrequency feedback signal generated by the receiving piezotransducer when affected by the acoustic signal after reverberation in the skull. This signal is then time-reversed and used to send a second signal towards the stimulator. This signal is inherently focused on the site of the stimulator and is configured to carry sufficient energy to operate the stimulator for DBS purposes.

The invention relates to a method and system for use in directly adjusting the timbre of a reproduced audio signal in any closed or partially enclosed space according to the measured reverberation time or other function describing the decay of sound within the space. The measurement of the reverberation time and the correction of the timbre are performed by a system that can be incorporated within the installed audio reproduction system, although a separate measuring system could alternatively be used. The measurement of decay time or reverberation time for the space is by known methods. The invention centres around the calculation and application of a correction filter determined directly from the measured decay time or reverberation time for the space.

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

Some embodiments of the present invention relate to a method and a circuit for processing an audio signal. In accordance with some embodiments of the present invention, there is provided an audio processing circuit including a first signal path, a second signal path and an output adder. The first signal path may be configured to allow an input signal to pass through the audio processing circuit substantially unaffected. The second signal path may include a reverberation filter and a cross-talk cancellation filter adapted to receive an output of the reverberation filter directly or indirectly. The audio processing circuit may further include an output adder, and the output adder may be adapted to receive and combine the output of the direct signal path and the output of the second signal path.

New methods of ultrasound imaging are presented that provide images with reduced reverberation noise and images of nonlinear scattering and propagation parameters of the object, and estimation of corrections for wave front aberrations produced by spatial variations in the ultrasound propagation velocity. The methods are based on processing of the received signal from transmitted dual frequency band ultrasound pulse complexes with overlapping high and low frequency pulses. The high frequency pulse is used for the image reconstruction and the low frequency pulse is used to manipulate the nonlinear scattering and / or propagation properties of the high frequency pulse. A 1st method uses the scattered signal from a single dual band pulse complex for filtering in the fast time (depth time) to provide a signal with suppression of reverberation noise and with 1st harmonic sensitivity and increased spatial resolution. In other methods two or more dual band pulse complexes are transmitted where the frequency and / or the phase and / or the amplitude of the low frequency pulse vary for each transmitted pulse complex. Through filtering in the pulse number coordinate and corrections of nonlinear propagation delays and optionally also amplitudes, a linear back scattering signal with suppressed pulse reverberation noise, a nonlinear back scattering signal, and quantitative nonlinear scattering and forward propagation parameters are extracted. The reverberation suppressed signals are further useful for estimation of corrections of wave front aberrations, and especially useful with broad transmit beams for multiple parallel receive beams. Approximate estimates of aberration corrections are given. The nonlinear signal is useful for imaging of differences in tissue properties, such as micro-calcifications, in-growth of fibrous tissue or foam cells, or micro gas bubbles as found with decompression or injected as ultrasound contrast agent.

An apparatus for computing filter coefficients for an adaptive filter is disclosed. The adaptive filter is used for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal. The apparatus has: an echo decay modeling means for modeling a decay behavior of an acoustic environment and for providing a corresponding echo decay parameter; and computing means for computing the filter coefficients of the adaptive filter on the basis of the echo decay parameter. A corresponding method has: providing echo decay parameters determined by means of an echo decay modeling means; and computing the filter coefficients of the adaptive filter on the basis of the echo decay parameters.