Abstract: A voice enhancement device includes: a sound receiving unit configured to receive an audio signal; a vehicle state monitor unit configured to monitor a vehicle state; a noise estimation unit configured to estimate a noise component for each frequency component using a cumulative histogram created by accumulating frequency of power of the audio signal received by the sound receiving unit for each frequency component; and a voice enhancer configured to suppress the noise component for each frequency component estimated by the noise estimation unit in the received audio signal, wherein the noise estimation unit resets the cumulative histogram on the basis of a monitoring result of the vehicle state monitor unit.

Abstract: Methods, systems, and apparatus for monitoring a sound are described. An audio signal is obtained and the audio signal is analyzed to generate an audio signature. An object type is identified based on the audio signature and an action corresponding to the object type is identified.

Abstract: Embodiments of systems and methods are described for determining which of a plurality of beamformed audio signals to select for signal processing. In some embodiments, a plurality of audio input signals are received from a microphone array comprising a plurality of microphones. A plurality of beamformed audio signals are determined based on the plurality of input audio signals, the beamformed audio signals comprising a direction. A plurality of signal features may be determined for each beamformed audio signal. Smoothed features may be determined for each beamformed audio signal based on at least a portion of the plurality of signal features. The beamformed audio signal corresponding to the maximum smoothed feature may be selected for further processing.

Abstract: A signal processing apparatus for dereverberating a number of input audio signals, where the signal processing apparatus includes a processor configured to transform the number of input audio signals into a transformed domain to obtain input transformed coefficients, the input transformed coefficients being arranged to form an input transformed coefficient matrix, determine filter coefficients upon the basis of eigenvalues of a signal space, the filter coefficients being arranged to form a filter coefficient matrix, convolve input transformed coefficients of the input transformed coefficient matrix by filter coefficients of the filter coefficient matrix to obtain output transformed coefficients, and the output transformed coefficients being arranged to form an output transformed coefficient matrix.

Abstract: The invention relates to a background noise estimator and a method therein, for supporting sound activity detection in an audio signal segment. The method comprises reducing a current background noise estimate when the audio signal segment is determined to comprise music and the current background noise estimate exceeds a minimum value. This is to be performed when an energy level of an audio signal segment is more than a threshold higher than a long term minimum energy level, lt_min, which is determined over a plurality of preceding audio signal segments, or, when the energy level of the audio signal segment is less than a threshold higher than lt_min, but no pause is detected in the audio signal segment.

Abstract: An image and audio reproduction device obtains an image captured by an image capturing device and a plurality of acoustic signals whose arrival directions are respectively a plurality of locations in an image capturing scope of the image capturing device, and sets, as one group, arrival directions of acoustic signals with a high degree of similarity of amounts of characteristics on the basis of a degree of similarity of the amounts of characteristics between the acoustic signals that are from the arrival directions adjacent to each other. The image and audio reproduction device adds information to the image captured by the image capturing device so as to reproduce the image, the information representing a position of the group in an image capturing scope of the image capturing device.

Abstract: Provided is a measurement device that evaluates an acoustic device 100 including a vibrating element and configured to allow sound to be heard by vibration transmission. The measurement device includes: an ear model unit 50 including an ear model 51 that is molded after a human ear and an artificial cartilage unit 54 that is joined to the ear model 51; and a vibration detector 56 disposed in the ear model unit 50.

Abstract: A mobile terminal apparatus portable by a user includes a recording unit, a memory, an estimating unit, and an output unit. The recording unit records a sound that has been produced by another apparatus. The memory stores a cause of an unusual noise that occurs in the other apparatus and intensities of individual frequency components of the unusual noise in association with each other. The estimating unit estimates a cause of the sound that has been produced by the other apparatus, on the basis of intensities of individual frequency components of the sound recorded by the recording unit and content stored in the memory, the intensities being calculated from the sound. The output unit outputs information about the cause of the sound estimated by the estimating unit.

Abstract: In some embodiments, a method for preparing monophonic audio for transmission to a node of a teleconferencing system, including steps of generating a monophonic mixed audio signal, including by a mixing a metadata signal (e.g., a tone) with monophonic audio indicative of speech by a currently dominant participant in a teleconference, and encoding the mixed audio signal for transmission, where the metadata signal is indicative of an apparent source position for the currently dominant conference participant. Other embodiments include steps of decoding such a transmitted encoded signal to determine the monophonic mixed audio signal, identifying the metadata signal, and determining the apparent source position corresponding to the currently dominant participant from the metadata signal. Other aspects are systems configured to perform any embodiment of the method or steps thereof.

Abstract: In embodiments of the present invention, methods for reproducing an acoustic environment are described, which may include accessing a computer stored multi-dimensional sound profile of a first space, measuring a multi-dimensional sound profile in a second space, comparing multi-dimensional sound profile of the first space and the multi-dimensional sound profile of the second space, accessing the sound characteristics of an audio output device that will serve as the audio output device in the second space, and modifying the audio output of a media content of the audio output device taking into account the sound characteristics of the audio output device, wherein the modifying reduces the difference as determined in the comparing between the multi-dimensional sound profile of the first space and the multi-dimensional sound profile of the second space as output by the audio output device.

Abstract: Compounds of Formula I or a stereoisomer, tautomer or pharmaceutically acceptable salt thereof are provided, which are useful for the treatment of hyperproliferative, pain and inflammatory diseases. Methods of using compounds of Formula I or a stereoisomer, tautomer or pharmaceutically acceptable salt thereof, for in vitro, in situ, and in vivo diagnosis, prevention or treatment of such disorders in mammalian cells, or associated pathological conditions are disclosed.

Abstract: Systems, methods, and computer program products to perform an operation comprising receiving a request to create a clip of a media file, wherein the request specifies a time duration for the clip, computing a metric value for each of a plurality of segments of the media file, identifying, based on the plurality of metric values, a first segment of the plurality of segments of the media file, wherein the metric value for the first segment of the media file exceeds a threshold metric value, and generating the clip of the media file, wherein the clip of the media file includes at least the first segment of the media file.

Abstract: System, apparatus and method for determining semantic information from audio, where incoming audio is sampled and processed to extract audio features, including temporal, spectral, harmonic and rhythmic features. The extracted audio features are compared to stored audio templates that include ranges and/or values for certain features and are tagged for specific ranges and/or values. The semantic information may be associated with audio signature data Extracted audio features that are most similar to one or more templates from the comparison are identified according to the tagged information. The tags are used to determine the semantic audio data that includes genre, instrumentation, style, acoustical dynamics, and emotive descriptor for the audio signal.

Abstract: In a system and method for maintaining the spatial stability of a sound field a background noise estimate may be estimated for each of a first signal and a second signal. A first gain coefficient may be calculated responsive to the first audio signal and the background noise estimate of the first audio signal. A second gain coefficient may be calculated responsive to the second signal and the background noise estimate of the second signal. The gain coefficients may be calculated using one or more gain coefficient calculators. A common gain coefficient may be selected from one of the first gain coefficient and the second gain coefficient. The selected common gain coefficient may be one that results in a least amount of audio signal modification and may be applied to each of the first signal and the second signal.

Abstract: Techniques for automatically connecting to a service controller are described herein. In one example, a service controller device includes a processor and a computer-readable memory storage device storing executable instructions that cause the processor to broadcast at least one of an access credential, connection information or an access credential hash embedded in an audio signal. The processor can also authenticate a client device based on a transmission of at least one of the connection information, the access credential, or the access credential hash from the client device to the client connector and transmit data to the client device in response to authenticating the client device.

Abstract: A filter for equalizing the frequency response of loudspeaker systems includes at least one band filter section (11) comprised an n-order high boost or cut shelving fitter (13) having a break point frequency, ?1, and an n-order low boost or cut shelving filter (15) having a break point frequency, ?2, wherein ?1<?2. The order, n, of at least one, and preferably both of the shelving filters of the band filter sections can be selected for adjusting the slope of the shelving filter at one or both of its break point frequencies. The high and low n-order shelving filters forming the band filter sections have substantially the same gain and produce a resultant band gain for the band filter section. Gain correction is provided for the selectable n-order high shelving filter and n-order low shelving filter for correcting the resultant band gain to a base gain level.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: A processing circuit may include: (i) an adaptive filter having a response that generates an anti-noise signal from a reference microphone signal, wherein the response is shaped in conformity with the reference microphone signal and a playback corrected error, and wherein the playback corrected error is based on a difference between an error microphone signal and a secondary path estimate; (ii) a secondary path estimate filter configured to model an electro-acoustic path of a source audio signal and having a response that generates a secondary path estimate from the source audio signal; (iii) a secondary coefficient control block that shapes the response of the secondary path estimate filter in conformity with the source audio signal and the playback corrected error by adapting the response of the secondary path estimate filter to minimize the playback corrected error; and (iv) a noise injection portion for injecting a noise signal into the source audio signal, wherein the noise signal is shaped based on the

Abstract: A personal safety system detects imminent danger or other event of interest and then modifies a sound panorama to focus a user's attention towards the direction of the danger or other event of interest. The personal safety system may isolate sounds originating from the direction of the danger or other event of interest, collapse the sound panorama towards the direction of the danger or other event of interest, or compress the sound panorama to align with the direction of the danger or other event of interest. The personal safety system may be integrated into an automobile or a wearable system physically attached to the user. Advantageously, the attention of the user may be drawn towards imminent danger or other events of interest without introducing additional sensory information to the user, thereby reducing the likelihood of startling or distracting the user.

Abstract: A method for allowing multiple audio sources to play sound through an audio speaker system is disclosed. The method includes designating a plurality of audio sources to provide audio sounds to the audio speaker system. The audio speaker system may receive an audio sound from each of the plurality of audio sources and create a playlist of received audio sounds. Creating the playlist may include ordering the received audio sounds in a sequential order and rotating between each of the plurality of audio sources. The audio speaker system may play each audio sound from the playlist via the audio speaker system.

Abstract: Provided is a loudspeaker system capable of easily determining control parameters for controlling the directional characteristics of a speaker set including two or more speaker units. The loudspeaker system is configured to include: a speaker set 11 including two or more speaker units 2; a camera 15 adapted to photograph an acoustic space of the speaker set 11; a monitor 22 adapted to display a camera image 30 photographed by the camera 15; and a directional control parameter generating part 24 adapted to generate directional control parameters providing the directional characteristics of the speaker set 11.

Abstract: A computing device determines responsiveness of a user to notification sounds output by the computing device, by determining whether for each notification sound the user acknowledged a notification corresponding to the notification sound on the computing device. The computing device can determine characteristics of the notification sounds output by the computing device. The computing device recommends suggested notification sounds that promote user acknowledgment to notifications, for use in relation to future notifications. The computing device may determine other notification sounds that share the characteristics of the notification sounds for which the user acknowledged the notifications corresponding to the notification sounds on the computing device. The suggested notification sounds can include the other notification sounds, and/or the notification sounds output by the computing device for which the user acknowledged the notifications corresponding to the notification sounds on the computing device.

Abstract: An automatic content recognition system that includes a user device for the purpose of capturing audio and generating an audio signature. The user device may be a smartphone or tablet. The system is also capable of determining sound level at a user device and refraining from audio monitoring and/or generating audio signatures when the sound level is below a threshold. Sound level may also be used to reduce the frequency of audio monitoring and/or signature generation. The system may have a database within the user device or the user device may communicate with a server having a database that contains reference audio signatures.

Abstract: A method for calculating a mean opinion score (MOS) during an ongoing Voice over Internet Protocol (VoIP) call is provided. The method may include determining a time delay between a VoIP source and a VoIP destination connected by a communications network. A start recording message is sent from the VoIP source to the VoIP destination. A first recorded call sample from the VoIP source and a second recorded call sample from the VoIP destination are generated, whereby the first and the second recorded call sample are generated with a recording delay value corresponding the determined time delay for synchronizing the first and the second recorded call sample. Using an intrusive call quality measurement, a first MOS value is calculated based on the first and the second recorded call sample. Using a non-intrusive call quality measurement, a second MOS value is calculated based on the first MOS value.

Abstract: Methods and systems are provided for enhancing speech intelligibility in electronic devices. During outputting of acoustic signal via an electronic device, measurement of forces applied by user of the electronic device against the device (or enclosure thereof) may be obtained. The force measurements may be used to assess and/or estimate the listening intelligibility experienced by the user. Further, the force measurements may be used to control or adjust a listening intelligibility stage applied during generation and/or processing of the acoustic signals that are outputted via the electronic device. In some instances, an audio input, corresponding to ambient noise affecting intelligibility, may be obtained, and may be used to control or assist in controlling the listening intelligibility stage.

Abstract: Disclosed herein are systems and methods enabling hearing aid fitting by a non-expert consumer at home. The method in one embodiment involves delivering a sequence of test audio signals corresponding to natural sound segments to a non-acoustic input of a programmable hearing device in-situ, while allowing the consumer to adjust fitting parameters based perceptual assessment of hearing device output. The sound segments define a fitting soundscape representing a practical range of sounds within the normal human auditory range, with each sound segment corresponding to one or more fitting parameters of the programmable hearing device. The consumer is instructed to listen to the output of the in-situ hearing device and adjust controls related to corresponding fitting parameters. In one embodiment, the fitting system comprises a personal computer and a handheld device providing calibrated test audio signals and a programming interface.

Abstract: A method for calculating a mean opinion score (MOS) during an ongoing Voice over Internet Protocol (VoIP) call is provided. The method may include determining a time delay between a VoIP source and a VoIP destination connected by a communications network. A start recording message is sent from the VoIP source to the VoIP destination. A first recorded call sample from the VoIP source and a second recorded call sample from the VoIP destination are generated, whereby the first and the second recorded call sample are generated with a recording delay value corresponding the determined time delay for synchronizing the first and the second recorded call sample. Using an intrusive call quality measurement, a first MOS value is calculated based on the first and the second recorded call sample. Using a non-intrusive call quality measurement, a second MOS value is calculated based on the first MOS value.

Abstract: The techniques discussed herein may facilitate real-time computation and playback of a propagated signal(s) perceived at a listener location in a three-dimensional environment in response to reception of a desired anechoic signal at a source location in the three-dimensional environment. The propagated audio realistically accounts for dynamic signal sources, dynamic listeners, and effects caused by the geometry and composition of the three-dimensional environment. The techniques may parameterize impulse response(s) of the environment and convolve the anechoic signal with canonical filters at run-time in a manner that respects the parameters of the parameterized impulse response(s). The techniques also provide for real-time computation and playback of a propagated audio signal perceived at a listener location in a virtual three-dimensional environment responsive to generation of source audio signals generated at multiple source locations in the virtual three-dimensional environment.

Abstract: Systems and methods may provide for a headset including a housing and a speaker positioned within the housing and directed toward a region external to the housing such as, for example, an ear canal when the headset is being worn. The headset may also include an ear pressure sensor positioned within the housing and directed toward the same region external to the housing. In one example, a measurement signal is received from the pressure sensor, one or more characteristics of an audio signal are automatically adjusted based on the measurement signal, and the audio signal is transmitted to the speaker.

Abstract: An apparatus including a recording system for recording at least sound, the recording system including a microphone; a distance input adapted to have an initial distance between a source of the sound and the apparatus entered into the apparatus; and a controller connected to the microphone. The controller is configured to determine a suggested distance for a better sound recording by the recording system based, at least partially, on the initial distance input into the distance input.

Abstract: A computerized system and method for music practice and training to assist a musician to improve their skills in maintaining a tempo and to improve ear-hand coordination, eye-hand coordination and other motor skills while learning to play an instrument. The system receives user defined beats per minute (BPM), rhythms or short segments of a music piece and generates audio reference outputs and visual display of music notes as stimuli. The user responses to these stimuli are measured in time and in the force and intensity of the responses and provided as feedback to the user to improve their response time and intensity to stimuli and assists the user in modifying the responses to the output stimuli to reduce the time between the output stimuli and the user response and thus improve the user's timing accuracy while playing music.

Abstract: An information processing apparatus that reproduces left and right audio signals based on data acquired from a sound source; acquires left and right surrounding sounds collected by left and right microphones included in a headphone unit connected to the information processing apparatus; combines a sound corresponding to at least one of the left and right surrounding sounds with at least one of the left and right audio signals; and outputs the left and right audio signals combined with the sound corresponding to the at least one of the left and right surrounding sounds to left and right earphones of the headphone unit.

Abstract: Methods and systems provide a partitioned IP core and hierarchical power management to reduce power consumption and footprint size of an “always-on” pulse density modulation (PDM) sensor system. The IP core may be partitioned into a register transfer level (RTL) block and a firmware block. The RTL may include a first stage decimation filter, storage, and, optionally, a sound energy detector. The firmware block may include subsequent decimation filter(s) and sensor processing logic, e.g., a sound trigger algorithm. In operation, the firmware block may conserve energy by being in a power-off or power-saving mode by default. Responsive to a trigger by the sound energy detector, the firmware block may wake up, receive data from the RTL block, and process the data. The sound energy detector may output the trigger based on characteristics of the received sample such as signal strength, noise strength, and type.

Abstract: A low power sound recognition sensor is configured to receive an analog signal that may contain a signature sound. Sparse sound parameter information is extracted from the analog signal and compared to a sound parameter reference stored locally with the sound recognition sensor to detect when the signature sound is received in the analog signal. A portion of the sparse sound parameter information is differential zero crossing (ZC) counts. Differential ZC rate may be determined by measuring a number of times the analog signal crosses a threshold value during each of a sequence of time frames to form a sequence of ZC counts and taking a difference between selected pairs of ZC counts to form a sequence of differential ZC counts.

Abstract: A low power sound recognition sensor is configured to receive an analog signal that may contain a signature sound. The received analog signal is evaluated using a detection portion of the analog section to determine when background noise on the analog signal is exceeded. A feature extraction portion of the analog section is triggered to extract sparse sound parameter information from the analog signal when the background noise is exceeded. An initial truncated portion of the sound parameter information is compared to a truncated sound parameter database stored locally with the sound recognition sensor to detect when there is a likelihood that the expected sound is being received in the analog signal. A trigger signal is generated to trigger classification logic when the likelihood that the expected sound is being received exceeds a threshold value.

Abstract: Embodiments for measuring content coherence and embodiments for measuring content similarity are described. Content coherence between a first audio section and a second audio section is measured. For each audio segment in the first audio section, a predetermined number of audio segments in the second audio section are determined. Content similarity between the audio segment in the first audio section and the determined audio segments is higher than that between the audio segment and all the other audio segments in the second audio section. An average of the content similarity between the audio segment in the first audio section and the determined audio segments is calculated. The content coherence is calculated as an average, the maximum or the minimum of the averages calculated for the audio segments in the first audio section. The content similarity may be calculated based on Dirichlet distribution.

Abstract: Embodiments of systems and methods are described for determining which of a plurality of beamformed audio signals to select for signal processing. In some embodiments, a plurality of audio input signals are received from a microphone array comprising a plurality of microphones. A plurality of beamformed audio signals are determined based on the plurality of input audio signals, the beamformed audio signals comprising a direction. A plurality of signal features may be determined for each beamformed audio signal. Smoothed features may be determined for each beamformed audio signal based on at least a portion of the plurality of signal features. The beamformed audio signal corresponding to the maximum smoothed feature may be selected for further processing.

Abstract: A mobile electronic device and a sound playback method thereof are provided. The mobile electronic device includes a sensor, a speaker, and a controller coupled to the sensor and the speaker. The sensor detects whether the speaker is blocked or not. When the speaker is blocked, the controller multiplies a sound signal by a transfer function and then outputs the multiplied sound signal. The speaker plays the sound signal outputted by the controller. The transfer function changes the direction in which the speaker plays the sound signal.

Abstract: A method for calculating a mean opinion score (MOS) during an ongoing Voice over Internet Protocol (VoIP) call is provided. The method may include determining a time delay between a VoIP source and a VoIP destination connected by a communications network. A start recording message is sent from the VoIP source to the VoIP destination. A first recorded call sample from the VoIP source and a second recorded call sample from the VoIP destination are generated, whereby the first and the second recorded call sample are generated with a recording delay value corresponding the determined time delay for synchronizing the first and the second recorded call sample. Using an intrusive call quality measurement, a first MOS value is calculated based on the first and the second recorded call sample. Using a non-intrusive call quality measurement, a second MOS value is calculated based on the first MOS value.

Abstract: A method and apparatus for generating a noise reduced output signal from sound received by a first and second microphone arranged as a microphone array. The method includes transforming sound received by the first microphone into a first input signal and sound received by a second microphone into a second input signal and calculating, for each of the frequency components, a weighted sum of at least two intermediate signals calculated from the input signals by means of complex valued transfer functions and real valued Equalizer functions. The method includes a weighing function with range between zero and one, with quotients of signal energies of the intermediate functions as arguments of the weighing function, and generating the noise reduced output signal based on the weighted sum of the intermediate functions and based on the weighted sum of the first and second intermediate function at each of the frequency components.

Abstract: A method that uses a microphone array for spatially selective sound pickup during an audio-video recording session is described. An audio signal for the audio-video recording session is generated using a beamforming process from the microphone array in accordance with a sound pickup directivity pattern. Ambient sound pressure level of the audio-video recording session is monitored while generating the audio signal. The sound pickup directivity pattern of the beamforming process is automatically adjusted during the audio-video recording session as a function of the monitored ambient sound pressure level. Other embodiments are also described and claimed.

Abstract: A method for calculating a mean opinion score (MOS) during an ongoing Voice over Internet Protocol (VoIP) call is provided. The method may include determining a time delay between a VoIP source and a VoIP destination connected by a communications network. A start recording message is sent from the VoIP source to the VoIP destination. A first recorded call sample from the VoIP source and a second recorded call sample from the VoIP destination are generated, whereby the first and the second recorded call sample are generated with a recording delay value corresponding the determined time delay for synchronizing the first and the second recorded call sample. Using an intrusive call quality measurement, a first MOS value is calculated based on the first and the second recorded call sample. Using a non-intrusive call quality measurement, a second MOS value is calculated based on the first MOS value.

Abstract: Devices and methods that match audio signatures to programming content stored in a remote database are disclosed. In one aspect of an embodiment, audio that includes primary audio from a device that outputs media content to one or more users is analyzed, in order to identify a presence or absence of corruption, and an audio signature is generated for an interval of time. In an aspect of a further embodiment, content being watched by a user is identified using a query audio signature and a message indicating the presence or absence of corruption.

Abstract: A microphone having an embedded haptic device, along with an associated program product and methodology for controlling the haptic device. A microphone is disclosed that includes: a haptic device embedded therein capable of outputting a haptic response; and a haptic control system having an input system for receiving an external input associated with a use of the microphone, an analysis engine for analyzing the external input, and a haptic response controller for triggering the haptic response.

Abstract: Measurements of body sounds in the ear canal may be used for many applications. However, reliability is dependent on the ear canal being properly sealed to allow the body sounds to achieve a detectable level. An apparatus is therefore provided for determining a seal quality indication for a seal of an ear canal. An ear canal microphone (201) provides a microphone signal to an input (203) which is coupled to a circuit (205) for generating a first signal from the microphone signal. The first signal may be the same as the microphone signal. A circuit (209) then determines the seal quality in response to the frequency spectrum for the first signal. A frequency transformer (207) may perform a frequency transformation on the first signal to generate a frequency spectrum for the first signal and. The seal quality indication may specifically be generated based on a detection of a low frequency boost.

Abstract: A volume control system, that can easily check current volume values with respect to volume values settable in a plurality of audio output devices when a volume value representing the volume values of the audio output devices is displayed, is provided. A controller of the volume control system can be connected to the plurality of the audio output devices, and can control the audio output devices. The controller obtains the volume values of the audio output devices and predetermined upper limit volume values stored in the audio output devices, and displays percentages of the volume values with respect to the predetermined upper limit volume values as total volume values. Further, the controller displays values that are determined by multiplying the total volume values and percentages of the predetermined upper limit volume values with respect to prescribed values as the volume values of the audio output devices.

Abstract: Systems and methods may provide for using an audio front end of a mobile device to sampled audio from an audio signal during a first portion of a periodic detection window, and reducing a power consumption of one or more components of the audio front end during a second portion of the periodic detection window. Additionally, a determination may be made as to whether voice activity is present in the audio signal based at least in part on the sampled audio. In one example, the length of the first portion and the length of the second portion are defined by a duty cycle of the periodic detection window.

Abstract: A circuit for correcting the output of an audio level meter comprises input means for generating the square or the absolute value of an input signal, a low pass filter having a predetermined attack time and release time, and output means for converting the output signal from a linear scale to a logarithmic scale. The circuit further comprises a correction means to which an information about whether the input signal to the audio level meter was subject to squaring or converting into an absolute value at the input, as well as the attack and release time of the low pass filter, are supplied as input values, and which provides, at its output, a value representing the difference between the output of the audio level meter and the true signal power of the input signal.

Abstract: In a method for deep tagging a recording, a computer records audio comprising speech from one or more people. The computer detects a non-speech sound within the audio. The computer determines that the non-speech sound corresponds to a type of sound, and in response, associates a descriptive term with a time of occurrence of the non-speech sound within the recorded audio to form a searchable tag. The computer stores the searchable tag as metadata of the recorded audio.

Abstract: An apparatus and method for preventing an acoustic shock of a portable terminal are disclosed. The apparatus includes a transceiver gain determining unit including a table of gain values for sound signals in each of one or more call modes, configured to output a particular gain value corresponding to a particular call mode; a reference value determining unit configured to receive the gain value from the transceiver gain determining unit, monitor a change in the gain value, and determine a predetermined reference energy value based on the change in the gain value; and an output level adjusting unit configured to receive an input signal, compare the input signal with the predetermined reference energy value, and adjust the level of the input signal to output an output signal.