Abstract: An apparatus and method for improving the performance of a determination of whether an earphone is mounted to a portable terminal, such as a portable terminal supporting a 3.5 mm earphone plug, are provided. An apparatus for recognizing whether the earphone is mounted to the portable terminal includes a common ground terminal and a switching terminal of an earphone plug receptacle for electrically connecting to each other and the same pole of an earphone plug when the earphone plug is mounted in the earphone plug receptacle, and an external output managing unit for generating an earphone mounting notification signal through a detector.

Abstract: A speaker apparatus includes: a speaker operable to emit a sound; a receiver operable to receive an operation signal for operating an external apparatus; a relay section; a storage operable to store correspondence between operation information and operation signals that includes a first operation signal and a second operation signal; and a controller: when the operation signal received by the receiver is the first operation signal, operable to cause the relay section to transmit the operation signal received by the receiver to the external apparatus, and operate the speaker apparatus in accordance with operation information that corresponds to the first operation signal; and when the operation signal received by the receiver is the second operation signal, operable to prevent the relay section from transmitting the operation signal received by the receiver to the external apparatus.

Abstract: An apparatus for generating an enhanced downmix signal on the basis of a multi-channel microphone signal has a spatial analyzer configured to compute a set of spatial cue parameters having a direction information describing a direction-of-arrival of a direct sound, a direct sound power information and a diffuse sound power information on the basis of the multi-channel microphone signal. The apparatus also has a filter calculator for calculating enhancement filter parameters in dependence on the direction information describing the direction-of-arrival of the direct sound, in dependence on the direct sound power information and in dependence on the diffuse sound power information. The apparatus also has a filter for filtering the microphone signal, or a signal derived therefrom, using the enhancement filter parameters, to obtain the enhanced downmix signal.

Abstract: The present technology provides adaptive noise and echo reduction of an acoustic signal which can overcome or substantially alleviate problems associated with mistaken adaptation of speech and noise models to acoustic echo. The present technology carries out a multi-faceted analysis to identify echo within the near-end acoustic signal to derive an echo model. Echo classification information regarding the derived echo model is then utilized to build near-end speech and noise models. These echo, speech, and noise models are then used to generate one or more signal modifications applied to the acoustic signal to preserve the desired near-end speech signal and reduce the echo and near-end noise signals. By building near-end speech and noise models utilizing echo classification information, the present technology can prevent adaptation of the speech and noise model to the acoustic echo.

Abstract: There is provided a sound signal processing device, in which an observation signal analysis unit receives multi-channels of sound-signals acquired by a sound-signal input unit and estimates a sound direction and a sound segment of a target sound to be extracted and a sound source extraction unit receives the sound direction and the sound segment of the target sound and extracts a sound-signal of the target sound. By applying short-time Fourier transform to the incoming multi-channel sound-signals this device generates an observation signal in the time-frequency domain and detects the sound direction and the sound segment of the target sound. Further, based on the sound direction and the sound segment of the target sound, this device generates a reference signal corresponding to a time envelope indicating changes of the target's sound volume in the time direction, and extracts the signal of the target sound, utilizing the reference signal.

Abstract: A speaker device for hands-free operation of a mobile device includes a first assembly operable for wireless communication with a mobile device, and a second assembly with a receptacle for docking at least a portion of the first assembly. The first assembly includes a first speaker and a first microphone, while the second assembly includes a second speaker. When the first assembly is docked in the receptacle of the second assembly, the mobile device is connected in wireless communication with the second speaker, and when the first assembly is removed from the second assembly, the mobile device is connected in wireless communication with the first speaker.

Abstract: An engine harmonic enhancement system. The EHE system uses multiple parameters, such as engine load, gear, number of cylinders operating, and transmission ratio, to determine EHE gains to determine EHE gain. The EHE system determines a separate EHE gain for each harmonic.

Abstract: A method for reducing noise in a shared media session. An indication is received from one or more of the participants in the shared media session. If the received indication is a first indication that indicates a background noise is present in the shared media session, the following steps are performed: a first counter is incremented for each of the first indications received from one or more of the plurality of participants, it is determined whether a background noise is present in the shared media session if the first counter exceeds a first threshold, an the shared media session is selectively muted such that the background noise is reduced if the background noise is determined to be present in the shared media session.

Abstract: A binaural hearing aid system includes a first hearing aid and a second hearing aid, each of which comprising a processor that is configured to process the digital input signal in accordance with a signal processing algorithm into a processed digital output signal, the processor including a compressor for compensation of dynamic range hearing loss based on the signal level, wherein wireless data communication of signal parameter from one of the first and the second hearing aids is performed at a data transmission rate with a time period between consecutive transmissions of the signal parameter from the one of the first and second hearing aids that is longer than an attack and release time of at least one of the compressors.

Abstract: A delay unit (16) for a conference audio system (1) adapted to delay audio input signals for an adjustable time delay, thereby generating audio output signals, is proposed.

Abstract: Systems and methods are provided to create an acoustic map of a space containing multiple acoustic sources. Source localization and separation takes place by sampling an ultra large microphone array containing over 1020 microphones. The space is divided into a plurality of masks, wherein each masks represents a pass region and a complementary rejection region. Each mask is associated with a subset of microphones and beamforming filters that maximize a gain for signals coming from the pass region of the mask and minimizes the gain for signals from the complementary region according to an optimization criterion. The optimization criterion may be a minimization of a performance function for the beamforming filters. The performance function is preferably a convex function. A processor provides a scan applying the plurality of masks to locate a target source. Processor based systems to perform the optimization are also provided.

Abstract: An audio signal high frequency component controlled in terms of directivity is reproduced, or an audio signal high frequency component compensated in terms of frequency characteristic or controlled in terms of directivity is reproduced, such that the reflected sound comes from a direction in which the high frequency component is intended to be localized. The sound pressure in a seat where a desired localization effect is not provided due to the arrangement of speakers is compensated such that the interaural amplitude level in the seat is equal to that of another seat. Thus, an equivalent level of localization effect is provided in a plurality of seats, especially for an audio signal high frequency component, without significantly increasing the number of the speakers.

Abstract: A hearing aid system includes a first hearing aid configured to communicate with a second hearing aid, wherein the first hearing aid comprises a microphone and an A/D converter for provision of a digital input signal in response to sound signals received at the microphone, a processor that is configured to process the digital input signal in accordance with a signal processing algorithm into a processed digital output signal, the processor including a compressor, a D/A converter and an output transducer for conversion of the processed digital output signal to an acoustic output signal, and a transceiver for data communication with the second hearing aid, wherein, a gain of the compressor of the first hearing aid is controlled by a first compressor control signal with a value that is substantially equal to a value of a second compressor control signal controlling a gain of a compressor in the second hearing aid, whereby a sense of direction is maintained.

Abstract: According to an embodiment, a method includes amplifying a signal provided by a capacitive signal source to form an amplified signal, detecting a peak voltage of the amplified signal, and adjusting a controllable impedance coupled to an output of the capacitive signal source in response to detecting the peak voltage. The controllable impedance is adjusted to a value inversely proportional to the detected peak voltage.

Abstract: An audio signal processing apparatus does not drive a drive unit in a period during which an audio signal used for noise reduction processing is acquired, or does not stop a recording operation until the audio signal used for noise reduction processing is acquired.

Abstract: A method and an apparatus for recognizing an accessory of a portable terminal are provided. The apparatus includes an interface unit in which the accessory is mounted, a power supply for supplying microphone bias power for recognizing the accessory in the interface unit, a current detector for determining an output current of the power supply, and a controller for determining whether the output current of the power supply exceeds a preset reference current when mounting of the accessory is sensed, and for recognizing a type of the accessory according to the output current of the power supply when the output current of the power supply exceeds the preset reference current. The apparatus may automatically recognize a type of an accessory connected with an interface unit to improve the convenience for a user.

Abstract: Methods, apparatuses, systems, and software are disclosed for providing a phone or other audio communication device with a fixed-path acoustic echo cancellation feature that compensates for a fixed-path acoustic coupling caused by transmission of sound from one or more speakers to one or more microphones through, for instance, the phone body itself.

Abstract: A speaker array apparatus for outputting sound beams of a plurality of channels based on audio signals of the plurality of channels, includes a directivity control section that controls a directivity of at least one of the sound beams of the channels so as to generate one or more pairs of the sound beams of the channels having roughly the same directivity, and a frequency characteristic applying section that applies a frequency characteristic to the audio signal corresponding to one of the sound beams of the channels in the one or more pairs to change a sound image localization position of the one of the sound beams of the channels in the one or more pairs.

Abstract: An apparatus provides both clipping protection and signal level conservation while the system operates in the original width type. The apparatus includes a first shifting unit right shifting a first digital input signal to provide a first shifted signal; a second shifting unit right shifting a second digital input signal to provide a second shifted signal; a combiner combining the first shifted signal and the second shifted signal to provide a combined signal; a soft limiter soft limiting the combined signal by reducing some of the amplitudes of the combined signal to provide a soft limited signal; and a third shifting unit left shifting the soft limited signal to provide an output signal.

Abstract: This disclosure details the implementation of apparatuses, methods and systems for audio processing and transmission. Some implementations of the system are configured to provide a method for encoding an arbitrary number of audio source signals using only a small amount of (transmitted or stored) information, while facilitating high-quality audio playback at the decoder side. Some implementations may be configured to implement, a parametric model for retaining the essential information of each source signal (side information). After the side information is extracted, the remaining information for all source signals may be summed to create a reference signal from which noise information for the original source signals may be reconstructed. The reference signal and the side information form the new collection of information to be transmitted or stored for subsequent decoding.