Abstract: Earphone systems and methods for automatically directing ambient sound to an earphone device are provided. An ambient microphone signal from an ambient microphone proximate a sound isolating earphone or headset device is directed to a receiver within an earphone device according to mixing circuitry. The mixing circuitry is controlled by voice activity of the earphone device wearer. This enables hands-free operation of an earphone system to allow the earphone device wearer to maintain situation awareness with the surrounding environment. During detected voice activity, incoming audio content is attenuated while ambient sound is increased and provided to the earphone device. User voice activity is detected by analysis of at least one of an ear canal microphone signal or an ambient sound microphone signal.

Abstract: The present techniques provide methods and systems for recording micro-holograms on a holographic disk using a plurality of counter-propagating light beams in parallel. The parallel counter-propagating light beams overlap to form interference patterns on a data layer and over multiple data tracks in the holographic disk. Rotating the disk enables the parallel recording of micro-holograms over multiple data tracks, thus reducing recording time. Further, the illumination pattern may include illuminated spots and non-illuminated regions, such that each illumination spot may cover a relatively small fraction of the data layer plane, possibly controlling the depth spread of the recorded micro-hologram. In some embodiments, data in the parallel signal beams may be retrieved from a master holographic disk or may be modulated into the parallel signal beams.

Abstract: Example methods, apparatus, systems and articles of manufacture to implement down-mixing compensation for audio watermarking are disclosed. Example watermark embedding methods disclosed herein include determining respective attenuation factors for a plurality of audio bands based on energy values associated with down-mixed audio samples corresponding to a first audio channel of a multi-channel audio signal and a second audio channel of the multi-channel audio signal. Disclosed example watermark embedding methods also include embedding a watermark in audio samples of the first audio channel based on the attenuation factors.

Abstract: Disclosed herein, among other things, are systems and methods for improved telecommunication for hearing instruments. One aspect of the present subject matter includes a hearing assistance method. The method includes receiving a first signal from a first hearing assistance device, receiving a second signal from a second hearing assistance device, and processing the first signal and the second signal to produce an output signal for use in telecommunication. In various embodiments, processing the first signal and the second signal includes comparing the first signal and the second signal, and selecting one or more of the first hearing assistance device and the second hearing assistance device for use in telecommunication based on the comparison. According to various embodiments, processing the first signal and the second signal includes combining the first signal and the second signal algorithmically to produce the output signal.

Abstract: Coupling between a user's ear and a speaker of a mobile device may be determined by measuring an impedance of the speaker. When the user presses the mobile device against the user's ear, the speaker impedance changes as a result of loading in the speaker's acoustic radiation impedance. The speaker impedance change may be correlated with the force applied by the user to the mobile device. The measured speaker impedance may be provided as feedback to an adaptive noise cancellation (ANC) algorithm to adjust the output at the speaker. For example, when the mobile device is removed from the user's ear, the ANC algorithm may be muted.

Abstract: Digital signal processing for microphone partial occlusion detection is described. In one embodiment, an electronic system for audio noise processing and for noise reduction, using a plurality of microphones, includes a first noise estimator to process a first audio signal from a first one of the microphones, and generate a first noise estimate. The electronic system also includes a second noise estimator to process the first audio signal, and a second audio signal from a second one of the microphones, in parallel with the first noise estimator, and generate a second noise estimate. A microphone partial occlusion detector determines a low frequency band separation of the first and second audio signals and a high frequency band separation of the first and second audio signals to generate a microphone partial occlusion function that indicates whether one of the microphones is partially occluded.

Abstract: In one aspect, a microphone with closely spaced elements is used to acquire multiple signals from which a signal from a desired source is separated. The signal separation approach uses a combination of direction-of-arrival information or other information determined from variation such as phase, delay, and amplitude among the acquired signals, as well as structural information for the signal from the source of interest and/or for the interfering signals. Through this combination of information, the elements may be spaced more closely than may be effective for conventional beamforming approaches. In some examples, all the microphone elements are integrated into a single a micro-electrical-mechanical system (MEMS).

Abstract: A control signal filter 2 receives a sound source signal determined by a control frequency specified in accordance with the vibration/noise source that produces vibration/noise, and outputs a control signal. A filter coefficient update unit 4 updates coefficients of the control signal filter 2 in response to a sound source signal and an error signal. A disturbance detector 6 outputs a disturbance detection result in response to the error signal and an estimated secondary vibration/noise signal. An update controller 7 adjusts an update step of the filter coefficient update unit 4 in accordance with the disturbance detection result.

Abstract: An electroluminescent earphone with a bending-resistance and a high-brightness includes earphone wires having bending-resistant electroluminescent wires, earbuds, an electroluminescent wire driver and an audio inputting plug. Each bending-resistant electroluminescent wire includes: a plurality of central electrodes; a luminescent layer coated on outer surfaces of the central electrodes; a transparent conductive layer coated on an outer surface of the luminescent layer; an outer electrode made of a plurality of wires arranged on an outer surface of the transparent conductive layer; and a transparent plastic covering the outer electrode and the transparent conductive layer. The bending-resistant electroluminescent wire is parallel with an earphone audio wire or coils around an outer surface of the earphone audio wire. A colored transparent plastic covers the bending-resistant electroluminescent wire and the earphone audio wire to form the earphone wire having the bending-resistant electroluminescent wire.

Abstract: A noise cancellation signal is generated by generating an ambient noise signal, representing ambient noise, and generating a noise cancellation signal, by applying the ambient noise signal to an feedforward filter, where the feedforward filter comprises a high-pass filter having an adjustable cut-off frequency, and by applying a controllable gain. The noise cancellation signal is then applied to a loudspeaker, to generate a sound to at least partially cancel the ambient noise. An error signal is generated, representing unwanted sound in the region of the loudspeaker. The phase of the ambient noise signal is compared to a phase of the error signal, and the gain is controlled on the basis of a result of the comparison, taking account of a phase shift introduced by the high-pass filter when performing the comparison.

Abstract: Systems and methods for determining the operating condition of multiple microphones of an electronic device are disclosed. A system can include a plurality of microphones operative to receive signals, a microphone condition detector, and a plurality of microphone condition determination sources. The microphone condition detector can determine a condition for each of the plurality of microphones by using the received signals and accessing at least one microphone condition determination source.

Abstract: An apparatus generates outgoing data to be provided on an optical disk in a burst cutting area. The burst cutting area further comprises markings which cause a marking frequency spectrum when reading out the burst cutting area. The apparatus comprises a channel coder which receives processed data and supplies the outgoing data having an outgoing data frequency spectrum with suppressed DC-content. The apparatus further comprises a data processing device which generates the processed data to obtain an outgoing frequency spectrum wherein a frequency component causing interference with a low frequent component of the markings is suppressed or not present.

Abstract: Disclosed are method and apparatus of generating noise that effectively performs performance evaluation of electronic equipment in a real environment by generating the noise having a statistic characteristic of real noise in a laboratory environment, by estimating a noise parameter set by measuring the noise and generating random noise data using a random noise data generation function having a statistic characteristic corresponding to the noise parameter set to generate radio noise.

Abstract: A priming signal processing technique modifies digital audio recordings to reduce the stress experienced by a listener's auditory system. A priming signal may reduce the instantaneous stress experienced by the auditory system during sudden changes in signal energy. The priming signal may leverage the temporal auditory masking, such that pre-signal priming additions may not result in obvious differences in perceived sounds.

Abstract: A method for determining an audio precompensation controller for an associated sound generating system comprising a total of N?2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L?2 inputs for L input signal(s) and N outputs for N controller output signals, one to each loudspeaker. It is relevant to: estimate (S1), for each one of at least a subset of the N loudspeaker inputs, an impulse response at each measurement position; specify (S2), for each one of the L input signal(s), a selected one of the N loudspeakers as a primary loudspeaker and optionally a selected subset S including at least one of the N loudspeakers as support loudspeaker(s); select (S2) at least one loudspeaker pair, that is required to be symmetrical with respect to the listing position; and specify (S3), for each primary loudspeaker, a target impulse response at each measurement position.

Abstract: Methods and systems for detecting the presence and frequency of clipping in an audio signal are provided. A clipping detection algorithm detects the presence of hard and soft clipping using histograms with intervals of samples, rather than attempting to identify the clipping value. Therefore, it is not essential to the algorithm that there be a large number of bins. Furthermore, the bins may be non-uniformly distributed since the number of samples belonging to lower amplitudes is of little importance. The detection algorithm is also configured to determine the severity and/or perceptual effect of any clipping found to be present in the signal by calculating the ratio of clipped samples to non-clipped samples. Temporal information on the occurrence of clipping in the signal is also used to evaluate perceptual effect.

Abstract: A microphone arrangement with improved directional characteristics is provided with at least two microphones (100, 102) and a signal processing arrangement (105). The signal processing arrangement is provided with a first (108) and a second input (109) for receiving the microphone signals of the at least two microphones. The inputs (108,109) are coupled to signal inputs of a first (110) and a second (111) multiplication circuit. The multiplication circuits are provided with control inputs for receiving respective first and second control signals, and with signal outputs. A control signal generator (112) is provided for generating the first and second control signals for the multiplication circuits (110,111). An arrangement (114) for a power corrected summation is provided, having a first and a second input coupled to the outputs of the first and second multiplication circuit, respectively, and having an output.

Abstract: A loudspeaker system includes a housing having an upper housing unit, a lower housing unit, and a cylindrical passageway connecting the upper housing unit and the lowering housing unit such that the upper housing unit and the lower housing unit are in fluid communication. A driver mounting plate is secured within the lower housing unit and divides the housing into an upper compartment and a lower compartment. At least one of the side walls of the lower housing unit includes a side wall aperture linking the lower compartment to an external environment surrounding the housing. A driver is mounted to the driver mounting plate, the driver being positioned to fire into the lower compartment of the housing.

Abstract: A mounting system is provided to allow the attachment of an acoustic collector to a handheld electronic device such that sound is conducted directly to the device's microphone. A fitted case or band encloses part of the device and includes a tube running from the collector to the device's microphone. In certain embodiments, the tube may be embedded in the case, and a detachable mount may be provided to connect the collector to the case. The collector may be a stethoscope chestpiece, or an open air collector, such as a parabolic collector.

Abstract: A conference system is described that transfers audio signals/streams between a near-end computing system and a far-end computing system. The near-end system may produce a plurality of microphone beams capturing various areas of the near-end. The beams may include primary beams directed at targets of interest and secondary beams focused on sound sources away from the targets of interest. The secondary beams may be selectively used as de-correlating signals for de-correlating the primary beams. The mixing may be made using a scale factor to produce a new primary beam. When the new beam does not provide significant de-correlation and the original raw primary beams are already sufficiently de-correlated, the original raw primary beams may be transmitted to the far-end system.