Abstract: A headset terminal for speech applications includes a headband assembly, an earcup assembly and a power source assembly. Processing circuitry is positioned in at least one of the earcup assembly and the power source assembly and includes speech processing circuitry for recognizing and synthesizing speech. A radio communicates with a central system to process the activity information of the headset terminal user. A rotatable microphone boom assembly includes controls mounted on opposite sides of a rotation axis to maintain a consistent orientation on the boom assembly with respect to the head of a user. The boom assembly snaps together with the earcup assembly to rotate. The headband assembly includes at least one transverse band and a sliding arm coupled to the earcup assembly for dynamically adjusting the position of the earcup assembly. A latch of the power source assembly snaps into position to secure it with the assembly and slides between latched and unlatched positions to secure a battery.

Abstract: A surround sound system comprises a receiver (301) for receiving a multichannel spatial signal that comprises at least one surround channel. A directional ultrasound transducer (305) is used for emitting ultrasound towards a surface to reach a listening position (111) via a reflection of the surface. The ultrasound signal may specifically reach the listening position from the side, above or behind of a nominal listener. A first drive unit (303) generates a drive signal for the directional ultrasound transducer (301) from the surround channel. The use of an ultrasound transducer for providing the surround sound signal provides an improved spatial experience while allowing the speaker to be located e.g. to the front of the user. In particular, an ultrasound beam is much narrower and well defined than conventional audio beams and can accordingly better be directed to provide the desired reflections. In some scenarios, the ultrasound transducer (305) may be supplemented by an audio range loudspeaker (309).

Abstract: A method and apparatus for receiving and playing a signal in a radio receiver to suppress microphonic feedback are provided by alternately pitch shifting a received audio signal. The pitch of the received audio signal is alternately shifted up and then down, repeatedly over successive intervals of the audio signal, to produce a pitch swing signal which is then played over a speaker. The alternating pitch shifting prevents the buildup of regenerative feedback normally caused by acoustic vibrations coupling into the radio receiver.

Abstract: An audio signal processing device includes a processing unit for convoluting head-related transfer functions with audio signals of a plurality of channels, and the processing unit includes a storage unit for storing data of a double-normalized head-related transfer function by normalizing a normalized head-related transfer function obtained by normalizing a head-related transfer function in a state in which a dummy head or a person is present in a position of the listener with a transfer characteristic in a pristine state in which the dummy head or the person is not present, using a normalized head-related transfer function obtained by normalizing a head-related transfer function in the state in which the dummy head or the person is present with a transfer characteristic in the pristine state, and a convolution unit for reading the data from the storage unit and convoluting the data with the audio signals.

Abstract: Method, device and computer program product for processing audio signals at the device, the device comprising an audio output for outputting audio signals. Audio signals are received at a plurality of microphones of the device. A characteristic of at least one of the audio signals received by the plurality of microphones is measured. A beamformer applies beamformer coefficients to the received audio signals, thereby generating a beamformer output. An echo canceller is applied to the beamformer output, thereby suppressing, from the beamformer output, an echo resulting from audio signals output from the audio output, wherein an operating parameter of the echo canceller is controlled based on the measured characteristic of the at least one of the audio signals received by the plurality of microphones.

Abstract: A system may comprise an input circuit, one or more filter circuits, one or more common signal detection circuits, and a controller. The input circuit may receive combined-game-and-chat audio signals generated from a mixing together of a chat audio signal and game audio signals. The filter circuit(s) may filter a first of the combined-game-and-chat audio signals to generate a first vocal-band signal and a first non-vocal-band signal, and filter a second of the combined-game-and-chat audio signals resulting in a second vocal-band signal and a second-non-vocal band signal. The common signal detection circuit(s) may detect strength of a signal component that is common to the first vocal-band signal and the second vocal-band signal, and strength of a signal component that is common to the first non-vocal-band signal and the second non-vocal-band signal. The controller may automatically control a volume setting based on one or both of the detected strengths.

Abstract: Ambient audio event detection in a personal audio device headset provides for directive response to external audible events. Depending on the type of event, an alert may be issued, speech may be communicated to another device, program material may be interrupted and/or resumed with or without repositioning, and program material may be modified or selected for compatibility with, or to overcome, the ambient environment indicated by the detected event.

Abstract: The present technology minimizes undesirable effects of multi-level noise suppression processing by applying an adaptive equalization. A noise suppression system may apply different levels of noise suppression based on the (user-perceived) signal-to-noise-ratio (SNR). The resulting high-frequency data attenuation may be counteracted by adapting the signal equalization. The present technology may be applied in both transmit and receive paths of communication devices. Intelligibility may particularly be improved under varying noise conditions, e.g. when a cell phone user is moving in and out of noisy environments.

Abstract: An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Abstract: A down-mixing device is provided. 5.1-channel sounds are down-mixed into 4-channel sounds with high-quality multi-channel surrounding sound effect reappeared. A common-acoustic-pole and zero (CAPZ) model is used to rebuild frequency responses of original head-related transfer functions (HRTF). Direct sounds, reflection sounds and late reverberation in a spatial sound field are gathered. Thus, varieties of sounds become rich and 3-dimensional hearing experiences are enhanced with low cost for a 4-channel earphones system.

Abstract: An apparatus includes a support unit for a handheld media device. The support unit has a first surface and a second surface. The first surface is coupleable to a substantially planar object. The substantially planar object is configured for mounting in a vertical position. The second surface is sized to support the handheld media device when the handheld media device is placed thereon.

Abstract: The instant development relates to utilizing the sonic motion generated by a speaker to move objects in various directions in response to the variation in the frequency and amplitude of the sonic vibrations. This can be used to move objects in a linear and/or rotating manner.

Abstract: When a correlation between an L channel and an R channel of a reproduction-target sound source is considerably high, virtual sound obtained from a reproduction signal is often more likely to be localized inside the head of a listener. A device includes: a correlation analysis unit (3) that analyzes a degree of correlation between a surround L channel signal (SL signal) and a surround R channel signal (SR signal); and an output signal control unit 4 that controls, based on the degree of correlation between the SL signal and the SR signal obtained as a result of the analysis performed by the correlation analysis unit (3), a ratio between: the signals outputted from a front L speaker (7) and a front R speaker (8); and the signals outputted from a near-ear L speaker (9) and a near-ear R speaker (10).

Abstract: Electronic devices and accessories for electronic devices such as headsets are provided. The electronic devices may produce audio output. The headsets may include earbuds with speakers that play the audio output for a user while the earbuds are located in the user's ears. Circuitry in an electronic device and a headset may be used in evaluating how well the earbuds are sealed to the user's ears. In response to seal quality measurements, informative messages can be generated for the user, overall earbud volume may be increased, balance adjustments may be made to correct for mismatched balance between left and right earbuds, equalization settings may be adjusted, and noise cancellation circuitry settings can be changed. Electrical impedance measurements and acoustic measurements can be used in evaluating seal quality.

Abstract: Provided are a speaker and a speaker system. The speaker includes a main unit and a sub unit. The main unit emits a sound in a first direction. The sub unit is formed integrally with the main unit and emits a sound in a second direction different from the first direction.

Abstract: Disclosed herein are system, method and apparatus with environmental noise cancellation. The instant disclosure is particularly adapted to a receiver module having at least two inputs. The two inputs respectively receive a main audio portion and the audio with majority of environmental noise. The system firstly calibrates the audio signals to reduce the error caused by the difference between the two inputs. An adaptive beamforming technology and a speech extractor are respectively used to extract the environmental noise portion with less main audio and the main audio portion with less noise. After a process of time-to-frequency domain transformation, a non-linear noise suppression technology is introduced into estimating the environmental noise and acquiring a gain. After noise suppression processed with the gain, a sequence of audio signals is output after a frequency-to-time domain transformation.

Abstract: Systems and methods are described by which microphones comprising a mechanical filter can be accurately calibrated to each other in both amplitude and phase.

Abstract: An active noise cancellation controller for performing noise attenuation in a system over a predetermined frequency rang. The controller comprises several components: a first input for a reference signal indicative of a noise level; a second input for receiving an error signal indicative of a remnant noise level; an output for providing a noise cancellation signal; a fixed feedback controller for processing the error signal; a fixed feed-forward controller for processing the reference signal; and an adaptive feed-forward controller having a digital adaptive finite impulse response filter arranged for operation on the reference signal the error signal.

Abstract: An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Especially those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Abstract: A frequency-domain upmix process uses vector-based signal decomposition and methods for improving the selectivity of center channel extraction. The upmix processes described do not perform an explicit primary/ambient decomposition. This reduces the complexity and improves the quality of the center channel derivation. A method of upmixing a two-channel stereo signal to a three-channel signal is described. A left input vector and a right input vector are added to arrive at a sum magnitude. Similarly, the difference between the left input vector and the right input vector is determined to arrive at a difference magnitude. The difference between the sum magnitude and the difference magnitude is scaled to compute a center channel magnitude estimate, and this estimate is used to calculate a center output vector. A left output vector and a right output vector are computed. The method is completed by outputting the left output vector, the center output vector, and the right output vector.