Abstract: An audio processing method and apparatus are described. In one embodiment, at least one first sub-band of a first audio signal is suppressed to obtain a reduced first audio signal with reserved sub-bands; suppressing at least one second sub-band of the at least one second audio signal to obtain at least one reduced second audio signal with reserved sub-bands; and mixing the reduced first audio signal and at least one reduced second audio signal. Alternatively, a first spatial auditory property is assigned to a first audio signal so that the first audio signal may be perceived as originating from a first position. Alternatively, rhythmic similarity between at least two audio signals is detected, and time scaling is applied to an audio signal in response to relatively high rhythmic similarity between the audio signal and the other audio signal(s); and then at least two audio signals are mixed.

Abstract: A method and system for improving echo suppression and/or duplexity of handsfree telephone applications is presented. An echo suppression unit for an electronic device comprising a loudspeaker and a microphone is described. The microphone is configured to capture a transmit signal, wherein the transmit signal comprises an echo of a receive signal rendered by the loudspeaker. The echo suppression unit is configured to determine, based on the receive signal, whether the receive signal comprises a first frequency component causing the echo of the receive signal to comprise a distortion component. The distortion component comprises one or more frequencies which are not comprised within the first frequency component. Furthermore, the echo suppression unit is configured to apply a post-filter to the transmit signal, if it is determined that the receive signal comprises the first frequency component. The post-filter is configured to selectively attenuate the distortion component.

Abstract: In accordance with embodiments of the present disclosure, a control circuit may be configured to, responsive to an indication to switch between gain modes of a signal path having an analog path portion and a digital signal path portion, switch a selectable analog gain of the analog path portion between a first analog gain and a second analog gain, switch a selectable digital gain of the digital signal path portion between a first digital gain and a second digital gain, wherein the product of the first analog gain and the first digital gain is approximately equal to the product of the second analog gain and the second digital gain, and control an analog response of the signal path to reduce the occurrence of audio artifacts present in the output signal as a result of the switch between gain modes of the signal path. A signal path may have an analog path portion and a digital signal path portion.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: A system and method for tracking and tracing motions of multiple incoherent sound sources and for visualizing the resultant overall sound pressure distribution in 3D space in real time are developed. This new system needs only four microphones (although more could be used) that can be mounted at any position so long as they are not placed on the same plane. A sample configuration is to mount three microphones on the y, z plane, while the 4th microphone on a plane perpendicular to the y, z plane. A processor receives signals from the microphones based on the signals received from noise sources in unknown locations, and the processor determines the locations of these sources and visualizes the resultant sound field in 3D space in real time. This system works for broadband, narrowband, tonal sound signals under transient and stationary conditions.

Abstract: Apparatuses, systems and methods for detecting corona using audio data are disclosed. A method of processing audio data to detect corona includes determining an indicator of energy at a substantially fundamental frequency in the audio data, determining an indicator of the energy at a plurality of harmonic frequencies in the audio data, determining an indicator of the noise energy in the audio data, determining a comparison indicator of the noise energy relative to the energy at the harmonic frequencies, determining a masking indicator having thresholds for each of the harmonics relative to the fundamental frequency, and determining a corona detection indicator. An apparatus for detecting corona includes a memory, an audio detector configured to obtain the audio data near an electrical conductor in an AC system, and a processor to process the audio data, fundamental and harmonic frequencies of the audio data and their corresponding thresholds to detect corona.

Abstract: The present disclosure is directed to devices, methods and systems for microphone arrays wherein enhancing performance of directional microphone arrays is provided. Enhanced performance of speaker phones is also provided. In certain embodiments, the housing of the device is configured to support the at least three microphones and the loudspeaker in a substantially first orientation; and the at least three microphones and the loudspeaker are arranged in a spatial relationship such that appropriate phase and delay characteristics achieve a substantial null response in the at least three microphones and in the loudspeaker in a substantial vertical direction away from the substantially first orientation over a desired audible range of frequencies and the device is able to provide a response to sounds over a range of first oriented elevations.

Abstract: A power amplifier system is disclosed, the power amplifier system comprising two power supplies, four amplifier half-bridges arranged as two amplifier full bridges, and an audio processor for establishing control signals for the amplifier full-bridges, where the audio processor and one of the power supplies share a common ground potential, whereas the other power supply has a floating ground; wherein both the amplifier with common ground and the amplifier with floating ground have an amplifier interconnection output and an amplifier output, and wherein the amplifier interconnection outputs are connected, and a load is connected to the amplifier outputs. Further are disclosed amplification methods and PA systems correspondingly.

Abstract: An improved method and arrangement for altering musical dynamics of a sound S included in a sound signal is disclosed. The altering of the musical dynamics is performed by filtering and amplification of the sound signal. The filtering is performed by the use of a parametric equalizer, the parametric equalizer having a first gain G1 and a resonance frequency fr being related to a pitch frequency fp of said sound S. The amplification is performed by an amplifier amplifying the sound signal with a second gain G2, the second gain G2 being dependent on the first gain G1.

Abstract: Examples described herein involve calibration of a microphone of a network device based on a first signal detected by the microphone of the network device and a second audio signal detected by a microphone of the playback device. While the network device is positioned within a predetermined physical range of a microphone of a playback device, a microphone of the network device may detect a first audio signal. The network device may also receive data indicating a second audio signal detected by the microphone of the playback device. Based on data indicating the first audio signal and the data indicating the second audio signal, the network device may identify a microphone calibration algorithm, and apply the microphone calibration algorithm when performing a calibration function associated with the playback device. Similar functions may also be performed by a computing device, such as a server to coordinate calibration of the microphone of the network device.

Abstract: A headset, a circuit structure of a mobile apparatus, and a mobile apparatus are provided. The headset includes at least one earphone, a microphone, an audio plug, and a headset cable. The audio plug includes a connector having a plurality of contacts. The microphone and the at least one earphone is coupled to the corresponding contacts of the audio plug through the headset cable. The headset cable includes a ground line, a microphone line, at least one audio line, and an antenna for receiving a television broadcasting signal. The antenna is coupled to one of the contacts coupled to the ground line, the microphone line, and the at least one audio line. The one of the contacts to which the antenna is coupled further serves as a radio frequency contact to provide the television broadcasting signal and an audio broadcasting signal.

Abstract: A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.

Abstract: An augmented reality environment allows interaction between virtual and real objects. Multiple microphone arrays of different physical sizes are used to acquire signals for spatial tracking of one or more sound sources within the environment. A first array with a larger size may be used to track an object beyond a threshold distance, while a second array having a size smaller than the first may be used to track the object up to the threshold distance. By selecting different sized arrays, accuracy of the spatial location is improved.

Abstract: Disclosed are exemplary embodiments of plugs and related methods. In an exemplary embodiment, a plug generally includes a connector configured for insertion into a jack of an electronic device. The plug also includes an offset member whereby the plug is configured to fit snugly relative to a protective case of the electronic device.

Abstract: A built-in audio apparatus includes a speaker unit, a mode setting unit configured to selectively set the speaker unit to one of a first position and a second position, and a controller configured to control the speaker unit depending on the setting of one of the first position and the second position. The mode setting unit is configured to be disposed in a surface of a structure, and includes an opening for emitting sound outputted from the speaker unit.

Abstract: Based on transducer status input signals indicative of whether headphones housing respective transducers are engaged with ears of a listener, a processing circuit may determine whether the headphones are engaged with respective ears of the listener. Responsive to determining that at least one of the headphones is not engaged with its respective ear, the processing circuit may modify at least one of a first output signal to the first transducer and a second output signal to the second transducer such that at least one of the first output signal and the second output signal is different than such signal would be if the headphones were engaged with their respective ears.

Abstract: In general, techniques are described for coding of spherical harmonic coefficients representative of a three dimensional soundfield. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may be configured to store a plurality of spherical harmonic coefficients. The one or more processors may be configured to perform an energy analysis with respect to the plurality of spherical harmonic coefficients to determine a reduced version of the plurality of spherical harmonic coefficients.

Abstract: The systems, devices, and methods provided herein can be used with an audio source to automatically attenuate a signal from the audio source in response to information about a local environment. In an example, an attenuation network circuit can be used to provide automatic or “hands-free” adjustment of a headphone or headset volume level to compensate for changing background sound or noise levels. In relatively noisy environments, the attenuation network can provide little or no attenuation of an audio source signal, and in less noisy environments, the attenuation network can provide greater attenuation of the audio source signal.

Abstract: An audio system for reproducing audio signals comprising a signal processor, the signal processor having at least two processing channels and thereby the signal processor being configured to receive at least two separate audio signals and the signal processor configured to automatically select, in dependence upon the identity of each of the at least two audio signals assigned to the at least two processing channels, processing settings appropriate for each of the at least two audio signals assigned to the at least two processing channels that are for use by the signal processor in independently processing each of the at least two audio signals in each of the processing channels prior to combining said at least two audio signals and the audio system being configured to change the processing settings for a processing channel in response to a change in the identity of the audio signal assigned to that processing channel.

Abstract: A interface circuit for an acoustic transducer provided with a first detection structure and a second detection structure has: a first input and a second input; a first processing path and a second processing path coupled, respectively, to the first input and second input and supply a first processed signal and a second processed signal; and a recombination stage, which supplies a mixed signal by combining the first processed signal and the second processed signal with a respective weight that is a function of a first level value of the first processed signal. The first and second inputs receive a respective detection signal associated, respectively, to the first detection structure and to the second detection structure of the acoustic transducer; and an output stage the first processed signal, the second processed signal or the mixed signal, on the basis of a second level value of the first processed signal.