125results about "Reverberation time" patented technology

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.

To efficiently detect a standing wave generated in a room, a standing-wave detection apparatus for detecting a standing wave in a predetermined space, comprises: a sound-receiving unit adapted to receive a sound generated from a sound source arranged in the predetermined space; a storage unit adapted to store time series sound pressure level data acquired by the sound-receiving unit during movement along a path in the predetermined space; an adjustment unit adapted to adjust the time series sound pressure level data stored in the storage unit, based on an adjustment curve determined using a lower envelope of the time series sound pressure level data stored in the storage unit; and a detection unit adapted to detect an existence position of a standing wave in the predetermined space based on the adjusted time series sound pressure level data.

An apparatus for determining a room dimension estimate comprises a receiver (101) providing an acoustic room response e.g. generated from acoustic measurements. A peak detector (103) detects a set of peaks in the acoustic room response in a frequency interval having an upper frequency of no more than 400 Hz. A store (107) comprises a set of peak profiles with associated room dimension data, and an estimator (105) determines the room dimension estimate from the associated room dimension data and a comparison of the set of peaks to the peak profiles. The estimator may perform the steps of first finding a matching peak profile for the set of peaks from the set of peak profiles; extracting first room dimension data associated with the matching peak profile(s) from the store; and determining the room dimension estimate in response to the first room dimension data. The peak profiles may represent calculated Eigenfrequencies.

An electronic device is provided. The electronic device includes a housing, at least one ultrasonic wave generator and at least one ultrasonic sensor operatively disposed on the housing, and a processor electrically connected to the at least one ultrasonic wave generator and the at least one ultrasonic sensor and configured to emit ultrasonic waves from at least one surface of the housing using the at least one ultrasonic wave generator, receive reflected waves reflected from at least one object through the at least one ultrasonic sensor, and determine a proximity between the electronic device and the object based on at least a difference between a first time, at which the ultrasonic waves are emitted, and a second time, at which the reflected waves are received.

The invention provides a method for generating an output indicative of acoustical sound transmission in a room. By using e.g. a point cloud representation of an acoustic environment, it is possible to calculate its acoustics from the interior information obtained from the depth camera. This approach is suitable e.g. for run-time applications since it is not based on an audible excitation that can disturb running audio. Also, the point-cloud model can be updated in real time according to the scene changes detected by depth-camera. This allows efficient acoustical simulation of dynamic, interactive environments. Although only geometrical information of a room is provided, high amount of surface details leaves possibility for implementation of material recognition algorithms that involve semantic mapping. This can provide information of reflective properties of surfaces or objects at a point level. Also, a high amount of details allows a good approximation of complex geometries, e.g. porous materials, and rough surfaces, thus a more natural simulation of wave phenomena like diffraction and scattering is possible.

Embodiments of the invention can relate to methods and devices for determining a room acoustic impulse response in a time domain. In one embodiment of a method, an acoustic input signal from an acoustic signal source is emitted into an acoustic room, an acoustic output signal is detected by an acoustic measuring device in the room and fed from the acoustic measuring device to an evaluating device and, by means of the evaluating device, from a reference signal corresponding to the acoustic input signal, and the acoustic output signal, if necessary after prior processing of the acoustic output signal, a room acoustic impulse response in the time domain of the acoustic room is calculated in real time and prepared for output in that, temporally in parallel and continuously, the acoustic input signal is emitted, the acoustic output signal is detected and, by means of the evaluating device, the room acoustic impulse response in the time domain is determined.

A song data downloading system and method of dispensing music are provided for use with a music data server and a musical jukebox. A song selection device and a control are provided. The control is operable to receive a user selection of particular song data with the song selection device. The user selection is made from compact disk song data or downloaded song data. The compact disk song data is from a compact disk that is in the compact disk storage and retrieval system of the jukebox. The downloaded song data is downloaded from a music data server. The control provides the selected song data to the music production system of the jukebox for producing audio from the selected song data. The control discards downloaded song data upon producing audio from the downloaded song data.

A method for determining the geometry and / or the localisation of an object comprising the steps of:sending one or more signals by using one transmitter;receiving by one or more receivers the transmitted signals and the echoes of the transmitted signals as reflected by one or more reflective surfacesbuilding by a computing module a first Euclidean Distance Matrix (EDM) comprising the mutual positions of the receivers;adding to the EDM matrix a new row and a new column, the new row and a new column comprising time of arrivals of said echoes and computing its rank or distance to an EDM matrixdetermining the geometry and / or the position of the object based on said rank or distance.

Disclosed are a method and apparatus for modeling a room impulse response. A method of modeling a room impulse response according to an embodiment of the invention includes receiving a sound pressure signal that is obtained by a microphone when an impulse-typed sound source is excited and detecting a room impulse response; determining boundaries between a plurality of intervals of the room impulse response such that the room impulse response is divided into the plurality of intervals on a time domain; and modeling the room impulse response using a different modeling method for each of the plurality of divided intervals. According to this invention, when a room impulse response is modeled, it is possible to decrease the number of parameters while maintaining accuracy.

The invention provides a deep sea seabed parameter inversion method based on seabed reverberation and transmission loss so as to effectively acquire seabed parameters in a deep sea environment. The method comprises the following steps: firstly, placing a single hydrophone in water; transmitting a sound source signal while a ship sails in an investigated and surveyed sea region; acquiring deep sea experimental data; extracting seabed reverberation and transmission loss experimental data; carrying out modeling and simulation on seabed reverberation model; matching the experimental data with a model simulated seabed reverberation time sequence; inverting to obtain sound velocity and density of the seabed; calculating transmission loss by using a ray model and matching the transmission loss with an experimental value; and inverting to obtain a seabed attenuation coefficient. The invention aims to provide an effective technical method for acquiring deep sea seabed parameters.

A method and apparatus implement time compression and expansion of audio data, with dynamic tempo change during playback. Dynamic changes in tempo are implemented at specific points in the audio signal corresponding to local minimums in the fade-in and fade-out characteristics of the compression / expansion scheme. An audio signal is marked to define temporal slices of audio data. Mark positions may be selected to minimize significant transient activity midway between consecutive marks. Fade-in and fade-out functions are associated with the leading side and trailing side, respectively, of each mark, creating a series of cross-fading “mounds” with peaks at each mark. When a tempo change is requested (e.g., a user selects a new tempo value in a user interface), the tempo change is delayed until the start of the next “mound” (i.e., the next fade-in). Thus, despite the tempo change, each mound uses a contiguous set of audio data, preventing the clicks and pops associated with skips in the audio data. Cross-fading minimizes any effects of desynchronization caused by overlapping mounds of differing speeds.

Measurement signals for calculating acoustic characteristics of the acoustic space are reproduced in a plurality of periods at sound reproduction intervals. A picked-up signal is acquired by picking up a reproduced signal. The picked-up signal is divided for each period, and the acoustic characteristics of the acoustic space are calculated from an arithmetic unit of the divided periodic signals and the measurement signal. Before sound reproduction, a background noise signal in the acoustic space is measured, and a peak frequency component of a frequency characteristic of the background noise signal is detected. The number of periods and sound reproduction intervals of measurement signals to be reproduced are controlled so as to cancel out a detected peak frequency component at the time of calculating arithmetic unit.

The invention relates to a reverberation time measurement method and particularly relates to an inversion method utilizing a single hydrophone to measure reverberation time of a non-anechoic pool. The method comprises steps that, a sound source level of a random non-directional sound source is measured in an anechoic pool by utilizing a calibrated hydrophone and is recorded and loaded to frequency and voltage amplitude on the sound source; the non-directional sound source and the hydrophone are put to a to-be-measured non-anechoic pool, frequency and voltage amplitude are recorded and are loaded to the non-directional sound source, the sound source and the hydrophone are simultaneously and slowly moved, and an average space sound pressure level is acquired by utilizing the hydrophone for measurement; the reverberation time of the non-anechoic pool can be acquired through inversion calculation according to the sound source level of the non-directional sound source in a free field and the measured average space sound pressure level of the sound source in the to-be-measured non-anechoic pool. The method has properties of high test efficiency and accurate result, and the average sound energy density attenuation time can be measured not on the basis of definition of the reverberation time.

Examples of the disclosure describe systems and methods for estimating acoustic properties of an environment. In an example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of a wearable head device, wherein the second audio signal is based on the second reverberation time.

The invention relates to a method and a device for detecting ultrasonic time of flight. The method is characterized in that an ultrasonic generator generates square wave ultrasonic pulse train excitation signals with 50 percent of duty cycle, the upper limit of the number of square pulses contained by the excitation signals is set when the wave shape of the received ultrasonic pulse train is still in a diamond shape; a timer is started up at the falling edge of the last square pulse of the excitation signal, received ultrasonic pulse train finds the peak point of maximum pulse of amplitude through differential amplification, gain control, high Q value band-pass filter, half wave rectification, variable index amplification, threshold comparison, differential and zero-crossing detection, and the timer is turned off at the peak point. The method not only achieves the high precision of phase difference method, but also overcomes the defects that the phase difference method can only be applied at short distances, breaks the tradition, and has broad application.

The present invention relates to a method for determining an acoustic property of an environment. According to the method a sound signal and at least two reflection signals of the sound signal (are received, and for each of the reflection signals a time difference between a time of reception of the sound signal and a time of reception of the respective reflection signal is determined. Furthermore, for each of the reflection signals, an angle between a reception direction in which the sound signal was received and a reception direction in which the respective reflection signal was received is determined. Based on the determined time differences, the angles and predetermined constraints defining arrangement constraints of the sound reflecting surfaces an arrangement of the sound-reflecting surfaces is determined.

A method and computer based program which performs a series of steps for automatically and accurately determining each note played in a song for each instrument and vocal. The method and program can transcribe or create sheet music for each individual instrument, as well as provide the ability to remove any combination of or individual instruments or vocal track basically from nearly any existing song, or future songs.

The invention provides a method for measuring and calculating the reverberation time in a vehicle. The method comprises a step S1 of calculating an original average value of the measured sound source sound pressure; a step S2 of setting a first value of the sound source sound pressure before attenuation and a second value after attenuation at each measuring point in the vehicle respectively according to the original average value of the sound source sound pressure; a step S3 of calculating the sound pressure attenuation slope at each measuring point according to the difference between the first value and the second value and the time of linear attenuation from the first value to the second value; and a step S4 of calculating the reverberation time of the measuring points according to the sound pressure attenuation slope at each measuring point. According to the invention, the average reverberation time in the vehicle is calculated according to the sound pressure attenuation slope, and in this way, the acoustical absorption coefficient of acoustic materials in the vehicle can be evaluated, so as to provide a basis for analyzing and improving the acoustic absorption property of the acoustic materials. In addition, the process is completed when the vehicle is in a static state, no frequent road test is required, the manpower and material resources and the cost are saved, and repeated verification and continuous improvement are carried out at all stages of development.

The invention discloses a measurement device of the underwater sound signal transmit time and a measurement method thereof and relates to the distance measurement and positioning in a limited space liquid yield environment. The method comprises the following steps: a first m-sequence generator generates an m-sequence; a first signal generator generates a first carrier signal, the m-sequence is used to modulate the first carrier signal and obtain a first modulation signal, the output and amplified modulation signal drives a transmitter underwater acoustic transducer to emit an underwater sound signal and also emit a synchronizing signal at the same time to trigger a signal collection circuit to work; a second m-sequence generator generates another m-sequence; a second signal generator generates a second carrier signal which is the same as the first carrier signal, the m-sequence is used to modulate the second carrier signal and obtain a second modulation signal; a receiver underwater acoustic transducer receives the underwater sound signal; the signal collection circuit processes the underwater sound signal; and a computer performs calculation to the second modulation signal and the treated signal to obtain the underwater sound signal transmit time. By adopting the method, the measurement accuracy of the transmit time can be increased.

A transmitting piezoelectric element and receiving piezoelectric elements and are arranged on a vibrating plate opposed to each opening of openings of a piezoelectric sensor, and, when an external force is applied in a vertical direction with respect to the vibrating plate opposed to the openings, a region which has a maximum principal stress which is a maximum value of the stress of the vibrating plate is a first region, and a region which has a minimum principal stress which is a minimum value of the stress of the vibrating plate is a second region, the receiving piezoelectric elements and are arranged in the first region and the transmitting piezoelectric element is arranged in the second region.

The invention provides a background noise energy eliminating measurement method of a reverberation time of a factory building. The method comprises the following steps of arranging a microphone in the factory building, giving out pulse response excitation through a sound source, and collecting a reverberation sound signal by using the microphone; selecting octave center frequency; filtering a frequency range corresponding to the collected signal, so as to obtain a sound pressure signal under the center frequency; choosing one segment of long enough background noise signals at a later-period stable stage to carry out energy integration, so as to obtain a noise energy average value per a unit time; choosing a long enough signal at a reverberation decay-period stage, so as to obtain an attenuation-period sound energy value after background noise energy is eliminated; linearly fitting a sound pressure level decay curve, so as to obtain a fitted gradient, and calculating the reverberation time through the gradient. By using the method, the influence of background interference noise on the measurement of the reverberation time can be removed; the method is applied to a factory building in which the background interference noise exits; the reverberation time of the factory building is measured.

The invention discloses a dereverberation control method and device for equipment with a microphone. The method comprises the following steps: using an audio signal played by the equipment, acquiring a reverberation parameter for indicating the environment reverberation degree of a room in which the equipment is located at any time; and dynamically adjusting a dereverberation mode used by the equipment according to a corresponding relationship between the reverberation degree indicated by the reverberation parameter at each time and a preset reverberation degree and the dereverberation mode. The scheme of the invention can significantly improve the recognition rate of the equipment to user voice through the dynamic dereverberation mode.

A system and associated method for estimating one or more reverberation parameters includes an adaptive, parametric, linear prediction filter having one or more power spectral density (PSD) estimates of signals of one or more channels as inputs. The prediction filter estimates at least one parameter related to reverberation time, such as T60, and can further estimate an additional parameter, such as Direct-to-Reverberant Ratio (DRR). The prediction filter may be adapted during a period of reverberation by minimizing a cost function. Adaptation can include using a gradient descent approach, which can operate according to a step size provided by an adaptation controller configured to determine the period of reverberation. One or more microphones can provide the signals. The reverberation parameters estimated can be applied to a reverberation suppressor, with an estimator that does not require a training phase and without relying on assumptions of the user's position relative to the microphones.

A device for detecting a resonant frequency comprises a sound source, a switch, a mixer, and a sound meter. The switch and mixer are capable of switching between a first state in which the switch and mixer output a measurement signal and a second state in which the switch and mixer output a synthesized signal containing the measurement signal and the signal output from a microphone. The device detects the resonant frequency based on comparison between the first amplitude frequency characteristic measured in the first state and the second amplitude frequency characteristic measured in the second state.

A method comprising determining a reverberation time estimate for an audio signal from a first part of an encoded audio signal representing the audio signal.

Disclosed is a method and system for diffraction-aware non-line of sight (NLOS) sound source localization (SSL) that may reconstruct an indoor space, may generate acoustic rays into the indoor space based on an audio signal collected from the indoor space, and may estimate a position of an NLOS sound source based on a point at which one of the acoustic rays is diffracted.

Load control systems and methods for shafts are provided. The load control system includes a sensor assembly. The sensor assembly includes a plurality of ultrasonic probes mounted to the shaft, each of the plurality of ultrasonic sensors configured to produce an ultrasonic wave on the shaft. The sensor assembly further includes a plurality of receivers mounted to the shaft, each of the plurality of receivers configured to sense the ultrasonic wave produced by one of the plurality of ultrasonic probes. The load control system further includes a controller communicatively coupled to the sensor assembly and configured to measure a travel time of the ultrasonic wave produced by each of the plurality of ultrasonic probes.

The invention relates to an ultrasonic wave transit time measuring circuit by secondary reflection waves. The ultrasonic wave transit time measuring circuit is used on an ultrasonic flowmeter and used for measuring the ultrasonic wave transit time between two sensors through the waveform similarity between primary received ultrasonic waves and secondary reflected ultrasonic waves. The ultrasonic wave transit time measuring circuit comprises a multi-path gating device (5), a programmable amplifier (6), waveform characteristic judging electrical systems (7-13), wherein the multi-path gating device (5) is used for ensuring that excitation signals, the primary received signals and the secondary reflected waves can accurately enter a subsequent circuit; the programmable amplifier (6) is used for processing different magnification time requirements of the primary received signals and the secondary reflected signals; the waveform characteristic judging electrical systems (7-13) are used for controlling the starting up / shutting down of a time counter (14) according to judgment results so as to complete the measurement of the transit time. According to the ultrasonic wave transit time measuring circuit, the energy source of the primary received ultrasonic waves is the same as that of the secondary reflected ultrasonic waves, so that a super-high similarity relation exists between the primary received ultrasonic waves and the secondary reflected ultrasonic waves. Thus, the problem that corresponding waveform characteristics are misjudged is avoided.