Abstract: A voice analyzer includes an apparatus body, a strap that is connected to the apparatus body and is used to hang the apparatus body from a neck of a user, a first voice acquisition unit provided in the strap or the apparatus body, a second voice acquisition unit provided at a position where a distance of a sound wave propagation path from a mouth of the user is smaller than a distance of a sound wave propagation path from the mouth of the user to the first voice acquisition unit, and an identification unit that identifies a sound, in which first sound pressure acquired by the first voice acquisition unit is larger by a predetermined value or more than second sound pressure acquired by the second voice acquisition unit, on the basis of a result of comparison between the first sound pressure and the second sound pressure.

Abstract: A howling detection device capable of detecting the occurrence of howling with a higher degree of accuracy is provided. The howling detection device includes a signal level calculating unit calculating, from the input signal, an input signal level being the signal level of an input signal at each given time, a level fluctuation estimated value calculating unit calculating, from the input signal level, a level fluctuation estimated value being a value gradually increasing or decreasing by a given amount with time, a threshold value calculating unit calculating, from the level fluctuation estimated value, a level threshold value changing in accordance with the level fluctuation estimated value, and a howling determination unit which determines that howling occurs in an input signal under the condition that the input signal level being higher than the level threshold value continues for a given time.

Abstract: Provided is a user authentication method including reproducing sound data of which a sound source in a first position of a space around a user is virtually localized using a Head-Related Transfer Function (HRTF) of the user toward the user, acquiring a second position of the space around the user, the second position being estimated by the user who has listened to the reproduced sound data as a position of the sound source; and authenticating the user according to a coincidence between the first position and the second position.

Abstract: Audio and video streams of a media file are transcoded just-in-time by keeping the transcoding process synchronized to a wall clock. The transcoding is initially configured slightly faster than real-time. The transcoded frames are pushed from the transcoder through a variable bandwidth connection as soon as they are transcoded. Transcoding is then monitored periodically to assess the bandwidth that was available during the transmission of a transcoded buffer. Two time intervals are measured: a buffer interval which is the difference in the transcoded buffer timestamps of the previous two transmissions, and a clock interval corresponding to the real time transmission time of the last transcoded buffer. If a significant difference is observed, the transcoder bit rate is adjusted by a factor that is derived from that difference.

Abstract: In one disclosed aspect, dynamic gain modifications are applied to an audio signal at least partly in response to auditory events and/or the degree of change in signal characteristics associated with said auditory event boundaries. In another aspect, an audio signal is divided into auditory events by comparing the difference in specific loudness between successive time blocks of the audio signal.

Abstract: A voice analyzer includes a first voice acquisition unit provided in a place where a distance of a sound wave propagation path from a mouth of a user is a first distance, plural second voice acquisition units provided in places where distances of sound wave propagation paths from the mouth of the user are smaller than the first distance, and an identification unit that identifies whether the voices acquired by the first and second voice acquisition units are voices of the user or voices of others excluding the user on the basis of a result of comparison between first sound pressure of a voice signal of the voice acquired by the first voice acquisition unit and second sound pressure calculated from sound pressure of a voice signal of the voice acquired by each of the plural second voice acquisition units.

Abstract: A method for impulsive noise mitigation using an adaptive blanker based on BPSK modulation system includes estimating narrowband Middleton parameters of impulsive noise from signals received for a fixed time period; calculating a threshold using the estimated narrowband Middleton parameters; operating the first blanker to suppress impulsive noise from the signal received at a given point of time through the comparison of the SNR value of the received signal and the threshold. Further, the method includes operating the second blanker to suppress impulsive noise from the received signal through the comparison of the summation of the threshold and the SNR value with an absolute value of the received signal, when the SNR value is above the threshold.

Abstract: Embodiments of the present invention provide a method and an apparatus for processing audio in video communication. The method includes: acquiring audio data and audio source position information corresponding to the audio data in the video communication, and performing longitudinal-depth processing on the audio data according to the audio source position information. According to the embodiments of the present invention, the audio data and the audio source position information corresponding to the audio data in the video communication are acquired first, and then the longitudinal-depth processing is performed on the audio data according to the acquired audio source position information to make it be audio data that provides a longitudinal-depth feeling that matches the audio source position information, so that sounds generated by objects at different front/back positions can be distinguished in the video communication.

Abstract: A method of decomposing a multichannel audio signal is described. The method includes, for each of a plurality of frequency components of a segment in time of the multichannel audio signal, calculating a corresponding indication of a direction of arrival. The method also includes, based on the calculated direction indications, selecting a subset of the plurality of frequency components. The method further includes, based on the selected subset and on a plurality of basis functions, calculating a vector of activation coefficients. Each activation coefficient of the vector corresponds to a different basis function of the plurality of basis functions.

Abstract: An audio-signal-processing circuit includes: a noise-detecting unit to detect presence or absence of noise in an audio signal generated based on an output from a tuner configured to receive a broadcast signal; a low-pass filter, having predetermined phase characteristics, to pass the audio signal having a band of frequencies lower than a predetermined frequency; a high-pass filter, having the predetermined phase characteristics, to pass the audio signal having a band of frequencies higher than the predetermined frequency; first- and second-output units to multiply the signals outputted from the low-pass and high-pass filters by first and second coefficients and output the multiplied signals, respectively; an adding unit to add the signals respectively outputted from the first and the second output units; and a coefficient control unit to, when the noise-detecting unit has detected the presence of noise, decrease the second coefficient below the first coefficient.

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

Abstract: An audio system includes a diagnostic capability to check for faults to a power source and faults to ground for output audio channels configured to drive loudspeakers. The fault to ground analysis involves analysis of a number of digital samples to determine if a predetermined threshold is exceeded during a predetermined period of time. The analysis may involve both a digital signal processor and a microprocessor performing a zero crossing analysis using the predetermined threshold and the predetermined window of time.

Abstract: In some embodiments, a method for upmixing input audio comprising N full range channels to generate 3D output audio comprising N+M full range channels, where the N+M full range channels are intended to be rendered by speakers including at least two speakers at different distances from the listener. The N channel input audio is a 2D audio program whose N full range channels are intended for rendering by N speakers nominally equidistant from the listener. The upmixing of the input audio to generate the 3D output audio is typically performed in an automated manner, in response to cues determined in automated fashion from stereoscopic 3D video corresponding to the input audio, or in response to cues determined in automated fashion from the input audio. Other aspects include a system configured to perform, and a computer readable medium which stores code for implementing any embodiment of the inventive method.

Abstract: A method of controlling audio recording using an electronic device and an electronic device are described. The electronic device comprises a microphone arrangement having a directivity pattern. A target direction relative to the electronic device is automatically determined in response to sensor data representing at least a portion of an area surrounding the electronic device. The microphone arrangement is automatically controlled in response to the determined target direction to adjust an angular orientation of the directivity pattern relative to the electronic device.

Abstract: Disclosed are an apparatus and a method for separating sound sources capable of learning distributions of corresponding sound sources based on the assumption that specific sound sources have specific distributions based on interchannel correlation parameter in audio signals providing space perception through a plurality of channels to separate an amount corresponding to energy contribution of the corresponding sound sources from mixture signals. Exemplary embodiments of the present invention can more precisely predict the channel distributions of the specific sound sources included in the input mixture signals and more accurately separate sound sources than a method for separating a sound source based on the channel according to the related art, under conditions that general channel distribution information of the specific sound sources are approximately modeled.

Abstract: Apparatus has at least one processor and at least one memory having computer-readable code stored therein which when executed controls the at least one processor to perform a method. The method comprises processing first and second (5) sets of substantially live audio data originating from first and second devices respectively to determine whether the first and second devices are observing a common audio scene, processing orientation information originating from the first and second devices to determine whether the devices are similarly oriented, and triggering a first action in response to determining both that the first and second (10) devices are observing a common audio scene and that the first and second devices are similarly oriented.

Abstract: A loudspeaker drive circuit has a microphone which forms part of an acoustic echo cancellation system. An input signal is processed before application to a loudspeaker driver, and the processing is controlled in dependence on the echo cancellation system performance, such as to control the extent to which the loudspeaker is driven into a non-linear operating region. In this way, the linearity can be controlled so as to provide an excursion limit, without needing a model of the loudspeaker or additional dedicated sensors.

Abstract: The present invention relates to an apparatus that senses the intensity of sounds and the velocity of objects. Specifically, the present invention relates to an apparatus that may prevent an audio signal from reaching a user, make a warning noise, or otherwise alert the user when a object is noisy and/or approaching the user at a threatening rate. More specifically, the present invention takes data from the surrounding area and compares it to one or more thresholds to determine an object's threat level. The present invention relates to one or more speakers, and includes, generally, one or more microphones to receive and measure the intensity of surrounding noises and one or more range sensors to detect distance, velocity, and acceleration.

Abstract: A system, method and program product for improved techniques for sound management and sound localization is provided. The present invention provides for improving sound localization and detection by inputting a predetermined location's dimensional data and location reference and processing detected sound details, detection device details and the associated location dimensional data as sound localization information for multi-dimensional display. The present invention provides mapping information of sound, people and structural information for use in multiple applications including residential, commercial and emergency situations.

Abstract: Provided is a method and apparatus for reproducing a three-dimensional (3D) sound field. A method of reproducing a 3D sound field may determine a control region based on a wavelength of an excitation frequency, set, based on a loudspeaker array, at least one candidate control point in the control region, and determine a non-uniform control point corresponding to a non-uniform loudspeaker.

Abstract: According to one embodiment, an audio controlling apparatus includes a first receiver configured to receive audio signal, a second receiver configured to receive environmental sound, a temporary gain calculator configured to calculate temporary gain based on environmental sound received by second receiver, a sound type determination module configured to determine sound type of main component of audio signal received by first receiver, and a gain controller configured to stabilize temporary gain that is calculated by temporary gain calculator and set gain, when it is determined that sound type of main component of audio signal received by first receiver is music.

Abstract: An audio interruption and buffering playback system includes a primary audio source for reproducing primary audio content and a secondary audio source for reproducing secondary audio content. A processing device detects an interruption event that includes the secondary audio source reproducing secondary audio content having priority over the primary audio source. The processor mutes the output of the primary audio content in response to the interruption event. A buffer buffers the audio content from the primary audio source during the interruption event. The buffered audio content is reproduced by the processor to the user at an accelerated playback speed following the interruption event.

Abstract: In some embodiments, a method for processing output of at least one microphone of a device (e.g., a headset) to identify at least one touch gesture exerted by a user on the device, including by distinguishing the gesture from input to the microphone other than a touch gesture intended by the user, and by distinguishing between a tap exerted by the user on the device and at least one dynamic gesture exerted by the user on the device, where the output of the at least one microphone is also indicative of ambient sound (e.g., voice utterences). Other embodiments are systems for detecting ambient sound (e.g., voice utterences) and touch gestures, each including a device including at least one microphone and a processor coupled and configured to process output of each microphone to identify at least one touch gesture exerted by a user on the device.

Abstract: To provide a speaker apparatus that can detect miswiring to a speaker unit or trouble of a speaker unit itself. The speaker apparatus is configured to include: two or more speaker units 11 that are arranged in a speaker housing 10; a sensor microphone 12 that is arranged in the speaker housing 10 and outputs a sound collection signal 6; a target unit selection part 20 that selects any one of the speaker units 11 as a target unit; a sound signal supply part 21 that supplies an external sound signal 4 to the target unit; and an error detection part 25 that provides an error output on the basis of the external sound signal 4 and the sound collection signal 6.

Abstract: An example information processing device determines a sound input to a microphone. The information processing device includes an obtaining section, a mean amplitude calculation section, and a determination section. The obtaining section obtains data of a sound detected by the microphone. For a sound of a predetermined determination segment, the mean amplitude calculation section calculates a mean amplitude, which is an average amplitude, for each of a plurality of partial segments included in the determination segment. The determination section determines whether or not the sound input to the microphone is a predetermined type of a sound (e.g., a sound made by breath blowing) based on the mean amplitudes for the partial segments.

Abstract: A device for measuring sound level, comprising: an inlet opening; a MEMS microphone for measuring sound level; and an external acoustic attenuator with a pressure divider comprising: a first branch between the inlet opening and the membrane of the MEMS microphone via an inlet channel and a resonant cavity; and a second branch between the resonant cavity and a vent chamber via a vent channel.

Abstract: A computer system is configured for performing audio matching. The computer system includes a client terminal and a server. The client terminal includes a classifier, a first landmark extraction module, a first encoder module and an integration module. The classifier is configured for estimating an identification difficulty. The first landmark extraction module is configured for determining the number of times for landmark extraction, and for extracting first landmarks. The first encoder module is configured for encoding the first landmarks. The server includes a second landmark extraction module, a second encoder module, a hash table, a table-lookup module, an analysis module and a decision module. The second landmark extraction module is configured for extracting second landmarks. The second encoder module is configured for encoding the second landmarks. The table-lookup module is configured for performing a table lookup. The decision module is configured for determining an audio matching result.

Abstract: A system can include a processor; memory operatively coupled to the processor; a microphone operatively coupled to the processor; a speaker operatively coupled to the processor; and circuitry that mutes the speaker based on detection of voices via the microphone. A method can include detecting two different voices via a microphone of a device; and responsive to the detecting, muting a speaker of the device. Various other apparatuses, systems, methods, etc., are also disclosed.

Abstract: Respective signal processing modules are executed in a DSP of an audio signal processing device, and also a signal abnormality detecting module is executed. The signal abnormality detecting module executes processing to detect a clip at metering points set at input and output ends of respective signal processing modules, and store in the storage an audio signal in which the clip is detected while correlating with the measuring points the metering point at which the clip is detected. The audio signal stored in the storage is outputted by the output module.

Abstract: A method and device are provided for detecting acoustic shocks in an audio stream. The method includes: breaking down the audio stream into audio frames; analyzing the audio frames in order to assign each audio frame a category value from among a plurality of predefined values; and determining the probability of an acoustic shock occurring in a current frame, based on a sequence of a given length of category values assigned to a set of frames, using a two-state Markov model, defined by a predetermined transition matrix and transmission matrix.

Abstract: Controlling a multimedia software application using high-level metadata features and symbolic object labels derived from an audio source, wherein a first-pass of low-level signal analysis is performed, followed by a stage of statistical and perceptual processing, followed by a symbolic machine-learning or data-mining processing component is disclosed. This multi-stage analysis system delivers high-level metadata features, sound object identifiers, stream labels or other symbolic metadata to the application scripts or programs, which use the data to configure processing chains, or map it to other media. Embodiments of the invention can be incorporated into multimedia content players, musical instruments, recording studio equipment, installed and live sound equipment, broadcast equipment, metadata-generation applications, software-as-a-service applications, search engines, and mobile devices.

Abstract: Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, when a user adjusts the audio settings of a playback device during the playback of a song or track, the user is warned if the currently playing song is “atypical” relative to other audio content. According to another embodiment, audio settings for a playback device are dynamically adjusted based on an audio profile of the audio that is played. According to yet another embodiment, audio settings are replicated onto other playback devices on a playback network.

Abstract: A method is provided in one example embodiment that includes monitoring a sound pressure level with an endpoint (e.g., an Internet Protocol (IP) phone), which is configured for communications involving end users; analyzing the sound pressure level to detect a sound anomaly; and communicating the sound anomaly to a sound classification module. The endpoint can be configured to operate in a low-power mode during the monitoring of the sound pressure level. In certain instances, the sound classification module is hosted by the endpoint. In other implementations, the sound classification module is hosted in a cloud network.

Abstract: An apparatus for evaluating sound quality and a method thereof may include generating an original sound, measuring a test sound generated from a subject of sound quality evaluation, analyzing a signal of the test sound measured according to a time domain, generating a plurality delay signals that are delayed from the signal of the time domain by one period, determining a degree of matching by matching the signal of the time domain to each delay signal, determining whether a white nose is occurred based on the degree of matching, and outputting the status of the white noise on the subject of the sound evaluation. The signal of the test sound may be analyzed according to a frequency domain, and a degree of occurrence of buzz and rattle is determined based on a fundamental frequency of the frequency domain and a plurality of Harmonics corresponding to the frequency domain.

Abstract: One embodiment of the present invention provides a system that uses a sound sensor to adjust the audio output of a device. During operation, the system uses the sound sensor to determine an ambient sound level for the environment in the proximity of the device. The system then adjusts a volume setting for the device adaptively based on the determined ambient sound level. Adaptively adjusting the volume setting allows the device to adapt to its audio environment and ameliorates potentially-disruptive audio outputs.

Abstract: An active noise control system for exhaust systems of a combustion engine operated vehicle comprises an anti-sound control connectable to an engine control of the vehicle and a loudspeaker connected to the control for receiving control signals and designed for generating an anti-sound in a sound generator, fluidically connectable to the exhaust system. In the control, at least two curves are stored in order to cancel airborne sound conducted in the exhaust system through outputting the signal to the loudspeaker. The curves cover different temperature ranges of the exhaust gas, which temperature ranges overlap one another by pairs or directly adjoin one another. The control is furthermore designed to select a curve suitable for a respective temperature of the exhaust gas conducted in the exhaust system from the available curves by means of signals output by the engine control and output signals to the loudspeaker making use of this curve.

Abstract: The invention relates to a method for determining a quality indicator representing a perceived quality of an output signal of an audio device with respect to a reference signal. Such audio device may for example be a speech processing system. In the method the reference signal and the output signal are processed and compared. The processing includes dividing the reference signal and the output signal into mutually corresponding time frames. The processing further includes scaling the reference signal towards a fixed intensity level. Time frames of the output signal are selected based on measurements performed on the scaled reference signal. Then, a noise contrast parameter is calculated based on the selected time frames of the output signal. A noise suppression is applied on at least one of the reference signal and the output signal based on the noise contrast parameter.

Abstract: An audio recording device comprises an audio input unit, a detect unit, and a control unit. The audio input unit includes a connection unit and an audio receive unit. The connection unit is configured to connect with an external pickup device. The audio receive unit is configured to receive the input of an audio signal from the pickup device. The detect unit is configured to detect the connection between the connection unit and the pickup device. The control unit is configured to display information indicating the input state of the audio signal from the pickup device in response to detection of the connection between the connection unit and the pickup device by the detect unit.

Abstract: Disclosed is a noise model generation device that generates a noise model suitable for each environment by determining whether an sound source of a detection target is included in sound information collected by an sound collector with high accuracy. The noise model generation device generates a noise model relating to noise information other than the sound source of the detection target included in the sound information collected by the sound collector, which acquires a power spectrum from the sound information; determines whether the sound source of the detection target is included in the sound information collected by evaluating a probability density distribution (histogram) of the power spectrum; and generates a noise model from the collected sound information when it is determined that the sound source of the detection target is not included in the collected sound information.

Abstract: Embodiments of the present invention provide a method and an apparatus for evaluating audio stream quality, where the method for evaluating audio stream quality includes: determining at least one non-silence audio data packet in a to-be-evaluated audio stream, where the to-be-evaluated audio stream includes at least one non-silence audio data packet, and each non-silence audio data packet includes at least one non-silence audio frame; and evaluating the at least one non-silence audio data packet in the to-be-evaluated audio stream to generate audio quality evaluation information. The method and the apparatus for evaluating audio stream quality provided by the embodiments of the present invention can avoid impact of a silence part on evaluation of audio quality, thereby improving accuracy of evaluating audio stream quality.

Abstract: An audio output apparatus and a method for audio correction is provided. An audio correction method of an audio output apparatus may include receiving a test signal from a terminal device, generating correction filter information of the terminal device based on the received test signal and a standard signal of the terminal device, and in response to an audio signal being received from the terminal device, correcting the received audio signal based on the correction filter information.

Abstract: A method for source selective real-time monitoring and mapping of environmental noise, comprising the steps of ? identifying sources in the area; ? designing places for the monitoring stations; ? creating a data collecting and data processing centre; ? measuring the noise impact, and observation of the sources with the sensors; ? defining the noise propagation; ? assemblying the monitoring stations including a sound measuring device, a computing unit and a communication unit; ? determining the resultant noise impact prevailing in the respective measurement points; ? determining the sources dominant in the individual moments on the basis of the data of the sensors; ? determining the average noise impact of the respective sources for a longer period; ? obtaining an average noise impact; ? extending the measurement into a noise impact map; ? calculating the effective noise impact of every noise source in every point of the area for every period; ? producing the effective noise impact map of the respective period

Abstract: The present invention relates to a method and a system for estimating acoustic noise levels. The invention may advantageously be applied in systems comprising multiple audio communication devices, such as e.g. telephones, mobile phones, headsets and headset base stations. It is an object of the present invention to provide respectively a method for estimating acoustic noise levels and a system for estimating acoustic noise levels, which enable a more efficient use of resources and may reduce cluttering in offices and other working environments than the prior art.

Abstract: A multifunctional wearable audio-sensing electronic device is disclosed in which audio is detected, detected audio is analyzed to determine characteristics of the audio, a warning is generated in response to determining that a decibel level of the detected audio is physically harmful, when the wearable audio-sensing electronic device is in a musical tuner mode, a deviation of the audio from a pre-determined audio characteristics of a musical instrument is displayed, when the wearable audio-sensing electronic device is in a hearing aid mode, an identification of the audio and a direction from which the audio originates is displayed, when the wearable audio-sensing electronic device is in an ambiance indicator mode, a visual representation of an audio ambience is displayed, and when the wearable audio-sensing electronic device is in an audio recorder mode, the audio is recorded to the memory and played back.

Abstract: The present invention relates generally to a portable/wearable device as well as application or a software program to be downloaded onto a mobile device or computer. The present invention is intended to measure sound intensity/level and provide an alert signal to remind user(s) or crowds (i.e groups of people) that they are speaking too loud or making too much noise. Furthermore, optionally, the device may play a pre-selected music or pre-recorded message when the sound level threshold is exceeded, which can encourage all of the users to use the bio-feedback of the present invention to relax when they see the visual que such as the light color changes, hear the music, message or produced from the device. Another object of the present invention is to help people to know whether his/her voice is loud enough to be heard by people who have hearing impairment.

Abstract: Earpieces and methods for acute sound detection and reproduction are provided. A method can include measuring an external ambient sound level (xASL), monitoring a change in the xASL for detecting an acute sound, estimating a proximity of the acute sound, and upon detecting the acute sound and its proximity, reproducing the acute sound within an ear canal, where the ear canal is at least partially occluded by an earpiece. Other embodiments are disclosed.

Abstract: An audio identification system generates a reference audio fingerprint associated with an event. The reference audio fingerprint is generated from samples of an audio signal associated with the event captured by multiple devices. To generate the reference audio fingerprint, fingerprints are generated from each sample, and the generated fingerprints are temporally aligned. Fingerprints associated a temporally overlapping portion of the audio signal are averaged, and the average value is associated with the temporally overlapping portion of the audio signal and included in the reference audio fingerprint. The reference audio fingerprint is stored along with identifying information, such as an event name, an event time, an event date, or other information describing the event associated with the audio signal from which the samples were captured.

Abstract: Disclosed is an apparatus for monitoring and/or recording noise levels at a location under surveillance over a time period. The apparatus includes a unit for capturing visual data of the location under surveillance, a unit for capturing audio data at the location under surveillance, a noise level process for deriving noise level data at the location under surveillance, and a unit to couple the visual and audio data with the noise level data.

Abstract: An image processing apparatus receives sound signals that are respectively output by a plurality of microphones, and detects a sound arrival direction from which sounds of the sound signals are traveled. The image processing apparatus calculates a sound level of sounds output from the sound arrival direction, and causes an image that reflects the sound level of sounds output from the sound arrival direction to be displayed in vicinity of an image of a user who is outputting the sounds from the sound arrival direction.

Abstract: A method including: determining a time difference between at least a first audio channel and a second audio channel of the same acoustic space; and enabling a corrective time shift between the first audio channel and the second audio channel when the time difference exceeds a threshold.