Abstract: An audio control system may receive a plurality of audio input signals. A switch or multiplexer may switch one of the plurality of audio input signals from respective audio sources to a power amplifier. A controller in communication with the multiplexer may control switching of the multiplexer. A display may provide the user with information regarding the switching of the audio signal inputs. A user interface may permit the user to manually switch the audio source. A signal detector in communication with an audio input signal may detect a signal from a priority source. When the signal from the priority source is detected, the controller may direct the multiplexer to route the signal from the priority source to the power amplifier.

Abstract: The invention relates to a method, computer, computer program and computer program product for speech quality estimation. The method comprises the steps of: determining a coding distortion parameter (QCOD), a bandwidth related distortion parameter (BW) and a presentation level distortion parameter (PL) of a speech signal; extracting a first coefficient (?l) and a second coefficient (?2), the first coefficient and the second coefficient being dependent on the coding distortion parameter; and calculating a signal quality measure (Q), where the signal quality measure is QCOD+?1BW+?2PL using the signal quality measure in a quality estimation of the speech signal.

Abstract: A method of operating a media player includes playing back audio media. During the step of playing back audio media, a maximum volume parameter is refined for the playing back of the media by the media player. The refining is based at least in part on the playing back of audio media during a time period prior to executing the maximum volume refining step. After a period of time, the maximum volume refining step is repeated. The refining is configured to prevent/minimize harm to hearing of the media player user based, for example, on the actual volume of media playback and time/duration profiles provided by occupational safety and/or other organizations.

Abstract: According to a preferred aspect, there is provided a system and method for controlled dynamics adaptation of audio works. The dynamics complexity of the audio is adapted depending on the listening environment and its dynamics properties. The instant invention utilizes a function of stepwise linear and logarithmical sections, which describe individual points of a curve and which therefore with the use of a parameter can be utilized to calculate and control the target value of the loudness range. The curve points are stored and provided as metadata of the musical program, which allows the dynamics scaling functionality to be implemented dynamically and not tied to a certain time or place.

Abstract: Systems and techniques for processing of media content information are described. Similarity information is determined for a plurality of media content segments captured by different devices that may be distributed through a space. The similarity information may define similarities between segments of media content overlapping in time. At least one transition pattern determines transitions between media content segments such as from an old content segment earlier in a timeline to a new content segment later in the timeline, with the new content segment being chosen based at least in part on similarity to the old content segment.

Abstract: An array speaker system canceling non-uniform radiation patterns and a method for implementing the same are provided. The method for canceling non-uniform radiation patterns includes predicting radiation patterns in an array speaker with respect to input signals, generating cancellation signals with respect to at least one region corresponding to non-uniform radiation patterns of the predicted radiation patterns, synthesizing the input signals and the cancellation signals, and outputting the synthesized signals to the array speaker. Sound signals from which non-uniform radiation patterns having distorted sound are cancelled such that a stable sound field having uniform radiation characteristics may be obtained and provided to a listener.

Abstract: A system and method for estimating a reverberation time is provided. The method includes estimating at least one room response of an audio capture environment with an acoustic echo canceller and generating an estimate of the reverberation time of the audio capture environment based on the at least one room response from the acoustic echo canceller.

Abstract: A system and method may be configured to process an audio signal. The system and method may track pitch, chirp rate, and/or harmonic envelope across the audio signal, may reconstruct sound represented in the audio signal, and/or may segment or classify the audio signal. A transform may be performed on the audio signal to place the audio signal in a frequency chirp domain that enhances the sound parameter tracking, reconstruction, and/or classification.

Abstract: A system and method that automatically disables and/or enables an acoustic beamformer is described herein. The system and method automatically generates an output audio signal by applying beamforming to a plurality of audio signals produced by an array of microphones when it is determined that such beamforming is working effectively and generates the output audio signal based on an audio signal produced by a designated microphone within the array of microphones when it is determined that the beamforming is not working effectively. Depending upon the implementation, the determination of whether the beamforming is working effectively may be based upon a measure of distortion associated with the beamformer response, an estimated level of reverberation, and/or the rate at which a computed look direction used to control the beamformer changes.

Abstract: A method for provisioning content to users, including for each of a plurality of items of media content, maintaining information about one or more owners of the item of media content, and maintaining information about one or more providers of the item of media content, for each of a plurality of users, maintaining information about items of media content acquired by the user and about player devices owned by the user, for each of a plurality of time periods, maintaining a history log of items of media content played by one or more of the plurality of users during the time period, receiving a request from a user to play a designated item of media content on a designated player device, during a current time period, identifying an appropriate provider of the designated item of media content, according to a location of the user and according to the designated player device, causing the identified provider of the designated item of media content to transmit the designated item of media content to the user's design

Abstract: An implantable hearing instrument is provided that reduces the response to unnaturally high vibrations (e.g., due to a patient's own voice), without necessarily or substantially affecting the response to desired signals. The implantable hearing instrument system is operative to selectively alter/lower the gain of signals that have a magnitude above a predetermined threshold. Signals having unnaturally large amplitudes (e.g., due to a patient's own voice) are amplified less than desired signals, so that a patient may experience a more representative sound.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes skipping extension information included in an input bitstream, extracting a three-dimensional (3D) down-mix signal and spatial information from the input bitstream, removing 3D effects from the 3D down-mix signal by performing a 3D rendering operation on the 3D down-mix signal, and generating a multi-channel signal using a down-mix signal obtained by the removal and the spatial information. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: A signal processing device includes: a sound adjustment amount calculation unit which calculates a sound adjustment amount for adjusting sound characteristics of each channel to a predetermined sound characteristic for each channel, using a sound signal that is obtained by collecting the outputs of each channel; an evaluation value calculation unit which calculates a coefficient allocation evaluation value for allocating a size of a filter coefficient necessary for the sound adjustment of the respective channels for each channel, based on the sound adjustment amount that is calculated by the sound adjustment amount calculation unit; and a filter coefficient calculation unit which calculates the filter coefficient for each channel using the coefficient allocation evaluation value that is calculated by the evaluation value calculation unit.

Abstract: An audio signal processing apparatus includes a tapping detection unit to which an audio signal picked up by a microphone is input and that detects a tapping input operation on the basis of an energy increase/decrease determination process for making a determination as to whether an increase and a decrease of an energy level of low-frequency components of the audio signal occur within a first time period; and a control unit that performs a certain control process that is set for a tapping input operation in response to a tapping input operation being detected by the tapping detection unit.

Abstract: Disclosed herein is an apparatus and method for measuring quality of audio. The apparatus includes a distorted signal generation unit, an extraction unit, a distortion level measurement unit, a distortion function generation unit, and a search and measurement unit. The distorted signal generation unit generates a plurality of distorted signals with respect to audio in compliance. The extraction unit extracts a fingerprint and AV information corresponding to the audio and fingerprints and AV information corresponding to the plurality of distorted signals. The distortion level measurement unit measures fingerprint distance differences, and arousal and valence (AV) distance differences. The distortion function generation unit generates a fingerprint distortion function and an AV distortion function.

Abstract: An acoustic management system configured to compensate for acoustical dampening is provided. The system includes a microphone configured to detect a first acoustic signal from an acoustic environment and a logic circuit. The logic circuit detects an onset of the acoustical dampening between the acoustic environment and the microphone. The logic circuit generates an acoustic damping compensation filter, which is applied to the first acoustic signal to generate a drive signal.

Abstract: An audio output controlling apparatus and method, and a mobile terminal system using the same control settings of an audio out port according to whether an audio frame includes an audio signal. The audio output controlling method includes receiving an audio frame, determining whether an audio signal is included in the audio frame, and setting, depending on whether the audio signal is included in the audio frame, a power voltage provided to an audio out port. The mobile terminal includes a wireless transceiver; an audio unit, through an audio out port, an audio frame among the wireless data; and an audio output controlling unit to control a power voltage of the audio out port. The apparatus includes a section determining unit; and a power controlling unit to set a power voltage provided to an audio out port according to whether the audio signal is included in the audio frame.

Abstract: The invention converts non audio systems into distributed audio sources for active noise control solutions. The system transforms non acoustic structures into soundboards using inertial type acoustic transducers. Acoustic parameters unique for each application due to the variation in properties of the sound board are compensated by equalizers. The invention also uses damping means to limit the reflection of bending waves from the edges. The inertial type acoustic transducer is driven by an amplifier. The acoustic signal to the amplifier is modified by a signal conditioner to compensate for the non optimal response of the acoustic system. An external controller communicates with the amplifier to control its operating parameters. A series of distributed audio sources in a variety of positions may each be addressable as a node on a network wherein noise detected at that source is analyzed and the system generates sound at that source to mask the noise.

Abstract: A wet road surface detection system is provided for a vehicle driving on a road. The wet road surface detection system includes a sound sensing device coupled to the vehicle for capturing noise of the vehicle tires as the vehicle drives on a road surface. A processor processes sounds captured by the audible sensing device. The processor quantifies a variable sound level of the captured noise between a first frequency and a second frequency. The processor determines whether each quantified sound level between the first frequency and the second frequency is above a predetermined sound level threshold below the sound level at the first frequency. The processor determines that the vehicle is driving on a wet road surface in response to each quantified sound level between the first frequency and the second frequency being above a predetermined sound level threshold below the sound level at the first frequency.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving the audio signal and preset information; obtaining preset matrix from the preset information, wherein the preset matrix indicates contribution degree of the object to output channel; and adjusting output level of the object by using the preset matrix. Accordingly, without user's setting for each object, if preset metadata to be applied to an audio signal is selected with reference to previously-set preset metadata, levels of objects included in the audio signal can be easily adjusted using preset rendering data corresponding to the selected preset metadata.

Abstract: When a communication terminal determines that restriction processing is to be applied based on a sound input through the communication terminal or a user instruction, the communication terminal restricts output of conference information received from a counterpart terminal.

Abstract: A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level.

Abstract: Waveform generating section 100 generates a waveform, and waveform collecting sections 200-1 to 200-3 collect the waveform generated by waveform generating section 100. Waveform comparing section 300 compares waveforms collected by waveform collecting sections 200-1 to 200-3 with each other. Group generating section 400 groups waveform collecting sections 200-1 to 200-3, which collected identical waveforms, into one group as a result of comparison.

Abstract: As the possible variations of “Hearing Thresholds”, “Hearing Loudness bandwidths” and “Voice Intonation” characteristics of people are finite, it is proposed to set a Database of these characteristics, where the data elements fully describe the Hearing and Talking characteristics of anyone, while many have the same characteristics. Thus any voice communication between two parties may be optimized by correcting the intensities of the call in the spectral domain, differently for each party and each ear. The optimizations are automatic given the “codes” of the parties and have a minimal latency. The system may be implemented either centrally in the world-wide-web or at the edges, in cellular phones, landline phones, VoIP, VoIM and in the audio parts of entertainment devices.

Abstract: Audio data to be played back is associated with meta data indicating the timing of beat that serves as a basis for a rhythm of a music piece. In a fast-forward playback mode, the associated meta data is read. A portion of the audio data containing the timing of beat is determined as a playback portion A and the remaining portion in a beat-to-beat interval C is determined as a non-playback portion B. Only the playback portion A is sequentially played back and output. The relative duration of a playback portion A and a non-playback portion B is determined according to a playback speed required in fast-forward playback.

Abstract: Digital signal processing microphone occlusion detection is described that can be used with a noise suppression system that uses two types of noise estimators, including a more aggressive one based on two audio signals (such as for non-stationary noises) and a less aggressive one based on one audio signal (such as for stationary noises). Decisions are made on how to select or combine the outputs of the noise estimators into a usable noise estimate, based on an occlusion function. The occlusion detection may alternatively be used to trigger an alert to users of multi-microphone audio processing systems, such as smart phones, headsets, laptops and tablet computers. Other embodiments are also described and claimed.

Abstract: The disclosed system and method for a mobile device combines information derived from onboard sensors with conventional signal processing information derived from a speech or audio signal to assist in noise and echo cancellation. In some implementations, an Angle and Distance Processing (ADP) module is employed on a mobile device and configured to provide runtime angle and distance information to an adaptive beamformer for canceling noise signals, provides a means for building a table of filter coefficients for adaptive filters used in echo cancellation, provides faster and more accurate Automatic Gain Control (AGC), provides delay information for a classifier in a Voice Activity Detector (VAD), provides a means for automatic switching between a speakerphone and handset mode of the mobile device, or primary microphone and reference microphones and assists in separating echo path changes from double talk.

Abstract: A method for equalizing audio and a video apparatus using the audio equalizing method are provided. The method for equalizing audio includes detecting the distance between a speaker mounted in a video apparatus and a reflective surface, and equalizing an audio signal to be output from the speaker based on the detected distance. Accordingly, attenuation of audio output is reduced, so audio output is optimized.

Abstract: Sound data acquisition means acquires, from a sound input device, sound data representing a sound signal. First evaluation means evaluates the sound signal represented by the sound data, by using a first criterion for detecting whether or not the sound inputted into the sound input device is a breath. Second evaluation means evaluates the sound signal represented by the sound data, by using a second criterion which is different from the first criterion.

Abstract: Signaturing methods and apparatus as described. In one example, a signaturing method includes capturing an audio signal, applying a first window function to a portion of the captured audio signal to produce a first windowed audio block, applying a second window function to the portion of the captured audio signal to produce a second windowed audio block, determining a first characteristic of a band of frequencies in the first windowed audio block, determining a second characteristic of the band of frequencies in the second windowed audio block, comparing the first characteristic to the second characteristic, and assigning a signature bit representative of the portion of the captured audio signal based on the comparison of the first characteristic and the second characteristic. Other examples are described.

Abstract: A test signal emitted by an earpiece is compared against the acoustic signal to determine if a sealing section of the earpiece is sealed properly. The degree of acoustic sealing is used to adjust the attenuation level of a sealing section of the earpiece or alert the user of the seal status of the sealing section.

Abstract: A spatial adaptation system for multiple-microphone sound capture systems and methods thereof are described. A spatial adaptation system includes an inference and weight module configured to receive a inputs. The inputs based on two or more input signals captured by at least two microphones. The inference and weight module to determine one or more weight values base on at least one of the inputs. The spatial adaptation system also including a noise magnitude ratio update module coupled with the inference and weight module. The noise magnitude ratio update module to determine an updated noise target based on the one or more weight values from the inference and weight module.

Abstract: An apparatus and processing method for assessing the electrical potential of cells in a living organism using a high-density sensor array having a density of at least about 9 electrodes per square inch for measuring the potential of the static and quasi-static electromagnetic fields of the group of cells proximal to each electrode, where the array can be placed overlying at least a portion of a surface of a living organism with substantially all of the electrodes electrically contactable with such surface, and the surface overlying a site containing cells of clinical interest.

Abstract: Systems and method for adapting a gain of an audio signal containing at least two different tracks with different signal level ranges. An example system includes a loudness determination unit for determining a perceived loudness of the audio input signal based on a psychoacoustic model of a human hearing. The system further includes a loudness adaptation unit configured to determine a loudness adaptation indicator based on the perceived loudness and a predetermined loudness, and to introduce the loudness adaptation indicator into the audio signal to generate an adapted gain signal. A gain determination unit is configured to adapt the gain for outputting the audio signal containing the at least two tracks based on the loudness adaptation indicators in the adapted gain signal at about the predetermined loudness.

Abstract: A noise alarm system includes a sound gathering circuit, a amplifying circuit, a wave filtering circuit, a comparing circuit, and a alarm circuit. The sound gathering circuit gathers a sound signal. The sound signal includes a noise signal and other signals. The amplifying circuit amplifies the sound signal. The wave filtering circuit filters the other signals and output the noise signal. The comparing circuit has a reference voltage and outputs the noise signal to the alarm circuit after comparing the noise signal with the reference voltage. The alarm circuit switches on when the noise signal greater than the reference voltage.

Abstract: In one embodiment, a pickup system includes a wind detector and a wind suppressor. The wind detector has a plurality of analyzers each configured to analyze first and second input signals, and a combiner configured to combine outputs of the plurality of analyzers and issue, based on the combined outputs, a wind level indication signal indicative of wind activity. The analyzers can be selected from a group of analyzers including a spectral slope analyzer, a ratio analyzer, a coherence analyzer, a phase variance analyzer and the like. The wind suppressor has a ratio calculator configured to generate a ratio of the first and second input signals, and a mixer configured to select one of the first or second input signals and to apply to the selected input signal one of first or second panning coefficients based on the wind level indication signal and on the ratio.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: A signal processing system for use in a multi-channel audio system having a plurality of power amplifier channels connected to a plurality of loudspeakers and power amplifiers, configured to receive and reproduce audio signals through the loudspeakers, and at least a first channel of the plurality of power amplifier channels amplifying a first audio signal, comprises a processor responsive to a signal level threshold applicable to at least said first channel, such that at and above the signal level threshold, the first audio signal in the first channel is amplitude limited and a portion of at least the first audio signal is mixed into at least a second channel. The amplitude limiting and signal mixing is configured so as to reduce introduction of at least one of: a) audible tonal distortion; and b) perceivable spatial distortion; of a sonic presentation due to said limiting and signal mixing.

Abstract: There is disclosed a microphone, a circuit, and a method. A microphone capsule may include an electret microphone and a field effect transistor (FET). A floating DC voltage source may have a first end connected to a drain terminal of the electret microphone capsule and a second end. A load resistor may be connected between the second end of the floating DC voltage source and a source terminal of the electret microphone capsule. A voltage follower may have an output connected to the source terminal of the electret microphone capsule and the first end of the floating DC voltage source. A coupling capacitor may couple an audio signal from the source terminal of the electret microphone capsule to an input of the voltage follower.

Abstract: A system which tracks a social interaction between a plurality of participants, includes a fixed beamformer that is adapted to output a first spatially filtered output and configured to receive a plurality of second spatially filtered outputs from a plurality of steerable beamformers. Each steerable beamformer outputs a respective one of the second spatially filtered outputs associated with a different one of the participants. The system also includes a processor capable of determining a similarity between the first spatially filtered output and each of the second spatially filtered outputs. The processor determines the social interaction between the participants based on the similarity between the first spatially filtered output and each of the second spatially filtered outputs.

Abstract: A method for a user to buy a song from a remote music source, the method comprising the steps of: the user inputting a song melody into an input device connected to the remote music source, wherein the user inputs the song melody using only their mouth; processing the tones of the input song melody so as to determine a set of tonal characteristics associated with the input tones; searching a music database for a song having a like set of tonal characteristics, so as to return at least one candidate song title to the user; playing a sample of the song associated with the at least one candidate song title for the user; permitting the user to purchase the song associated with the at least one candidate song title if desired; and delivering a purchased song to the user via a digital transmission medium.

Abstract: Method and arrangement for processing of a speech quality estimate, which involve adaption of a speech quality estimate based on information related to the bandwidth of a reference signal used when determining said speech quality estimate, such that the adapted speech quality estimate is independent of the bandwidth of the reference signal. The method and arrangement enable objective speech quality measurements or assessments to be performed on a unified bandwidth scale, independent of the bandwidth of a reference signal, which allows e.g. a more relevant comparison of communication systems and/or equipment, such as e.g. codecs.

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: Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract moving source signals from mixtures of source signals by way of independent component analysis. Source motion is modeled by direct to reverberant ratio in the separation process, and independent component analysis techniques described herein use multivariate probability density functions to preserve the alignment of frequency bins in the source separation process.

Abstract: According to one embodiment, a sound signal compensation apparatus includes an input module, a compensation module, and an output module. The input module receives identification information identifying a first frequency with regard to a resonance of an ear closed by an earphone or headphone. The compensation module performs first compensation emphasizing a second frequency on a sound signal, the second frequency being determined based on the identification information or the first frequency. The output module outputs the compensated sound signal. The compensation module is configured to perform the first compensation emphasizing the second frequency, at which emphasis is greater than or equal to 2 dB and less than or equal to 12 dB.

Abstract: Techniques for scene change detection around seed points in media data are provided. Media features of many different types may be extracted from the media data. One or more statistical patterns of media features in a plurality of time-wise intervals around a plurality of seed time points of the media data may be determined using one or more types of features extractable from the media data. At least one of the one or more types of features comprises a type of features that captures structural properties, tonality including harmony and melody, timbre, rhythm, loudness, stereo mix, or a quantity of sound sources as related to the media data. A plurality of beginning scene change points and a plurality of ending scene change points in the media data may be detected, based on the one or more statistical patterns, for the plurality of seed time points in the media data.

Abstract: In various embodiments, a system is used to provide an apparatus configured to measure sound in an ear canal of a wearer's ear at a distance from a tympanic membrane of the ear. The sound is measured and received by the apparatus to produce a signal. A frequency analysis is performed on the signal to determine output as a function of the frequency and to determine the frequency of the minima (null). Further, a distance equal to a quarter wavelength of the null frequency is calculated. A correction factor associated with the quarter wavelength is retrieved and applied to the output to generate a corrected output. An estimated sound pressure level at the tympanic membrane from the corrected output is produced.

Abstract: Example methods and systems for displaying one or more indications that indicate (i) the direction of a source of sound and (ii) the intensity level of the sound are disclosed. A method may involve receiving audio data corresponding to sound detected by a wearable computing system. Further, the method may involve analyzing the audio data to determine both (i) a direction from the wearable computing system of a source of the sound and (ii) an intensity level of the sound. Still further, the method may involve causing the wearable computing system to display one or more indications that indicate (i) the direction of the source of the sound and (ii) the intensity level of the sound.

Abstract: A mobile communication environment (100) can include a mobile device (160) to measure and send sound pressure level data. The mobile device (160) can initiate the collection of audio information responsive to detecting a trigger event. Mobile device (160) can measure or calculate the sound pressure level from the audio information. Metadata including time information and geographic location information can be captured with the collected audio information. Mobile device (160) can send the sound pressure level data and metadata through a wired or wireless communication path to a database (614).

Abstract: In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal, calculating the difference in spectral content between successive time blocks of the audio signal, and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold. In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events. Optionally, the invention may also assign a characteristic to one or more of the auditory events. Auditory events may be determined according to the first aspect of the invention or by another method.