Abstract: An audio output apparatus and a method are provided, which includes calculating a temperature value of a voice coil of the audio output apparatus using a heat transfer model algorithm, in response to power being supplied to the voice coil; adjusting an output level of an audio signal by determining a gain value to adjust the output level of the audio signal according to the calculated voice coil temperature value, and outputting the audio signal with the adjusted output level.

Abstract: An audio system has a first channel for receiving a first input signal and driving a first speaker and a second channel for receiving a second input signal and driving a second speaker. A first feedforward circuit couples an input of the second channel circuit to an input of the first channel circuit. A second feedforward circuit couples an input of the first channel circuit to an input of the second channel circuit. Circuit parameters of the first and the second feedforward circuits are determined such that a first detected output signal is zero when the first input signal is non-zero and the second input signal is zero, and a second detected output signal is zero when the second input signal is non-zero and the first input signal is zero. The audio system is configured to operate using the determined circuit parameters for the first and the second feedforward circuits.

Abstract: Methods and apparatus are disclosed for subwoofer calibration. An example method includes determining a crossover frequency of a playback system, and the playback system includes the subwoofer and a playback device. The example method includes the subwoofer and the playback device configured to output a multimedia content in synchronization. The example method also includes outputting from the subwoofer and the playback device a series of tones near the crossover frequency. The example method also includes prompting a user to select a tone set from the series of tones, and includes automatically adjusting a phase of the subwoofer in relation to the playback device based on the user selection.

Abstract: A MEMS microphone package is described that includes a first node and a second node having nearly equivalent 3D parasitic capacitances relative to a preamplifier of the microphone package, such that any noise generated on the first node is equivalent to any noise generated on the second node. An external power supply is connected to the first node and provides a bias voltage signal to the MEMS microphone package via the first node. An inverting amplifier is connected between the power supply and the second node. A third node is connected to the first node through a packaging parasitic capacitor, while the second node is connected to the third node through either an intended parasitic capacitor or an explicit capacitor. The noise coupled from the external power supply to the third node is then cancelled by summing the inverted power supply noise into the third node.

Abstract: A videoconferencing system has a videoconferencing unit that use portable devices as peripherals for the system. The portable devices obtain near-end audio and send the audio to the videoconferencing unit via a wireless connection. In turn, the videoconferencing unit sends the near-end audio from the loudest portable device along with near-end video to the far-end. The portable devices can control the videoconferencing unit and can initially establish the videoconference by connecting with the far-end and then transferring operations to the videoconferencing unit. To deal with acoustic coupling between the unit's loudspeaker and the portable device's microphone, the unit uses an echo canceller that is compensated for differences in the clocks used in the A/D and D/A converters of the loudspeaker and microphone.

Abstract: A method includes comparing a magnitude value of an audio signal in a fixed unit section with a target volume level, measuring audio gain in each fixed unit section using specific functions which are each determined to be different according to a ratio of the target volume level to the audio signal magnitude, and providing the measured audio gain to the audio signal in the fixed unit section.

Abstract: A method for reducing a disturbance on a signal path is provided. The disturbance is caused by a capacitance (203, 214)of a semiconductor switch (131-133) of an integrated circuit (130). The semiconductor switch (131-133) switches the signal path. The capacitance (203, 214) acts between the semiconductor switch (131-133) and a power supply terminal of the integrated circuit (130). According to the method, the power supply terminal of the integrated circuit (130) is coupled via an impedance (138, 139) to a power supply (137, 140).

Abstract: A discreet personal recording necklace and pendant system for recording an audio signal for playback is suspended around a neck of a user. The system features a hollow tube adapted for housing wiring, a cylindrical clasp located on the hollow tube adapted for housing a power supply, and a pendant located on the hollow tube. The pendant features a microprocessor located in a pendant cavity connected to the power supply via wiring through the hollow tube. A pendant first surface has a capacitive touch sensor and a microphone. A pendant second surface features a speaker. The microprocessor is operated via a voice activation component in the microprocessor. The audio signal is received via the microphone and emitted via the speaker. The discreet personal recording necklace and pendant system for recording the audio signal for playback is suspended around the neck of the user.

Abstract: A speech from a speaker proximate to one or more microphones within an environment can be received. The microphones can be a directional microphone or an omni-directional microphone. The speech can be processed to produce an utterance to determine the identity of the speaker. The identity of the speaker can be associated with a voiceprint. The identity can be associated with a user's credentials of a computing system. The credentials can uniquely identify the user within the computing system. The utterance can be analyzed to establish a zone in which the speaker is present. The zone can be a bounded region within the environment. The zone can be mapped within the environment to determine a location of the speaker. The location can be a relative or an absolute location.

Abstract: Various embodiments of systems and methods for reducing audio noise are disclosed. One or more sound components such as noise and network tone can be detected based on power spectrum obtained from a time-domain signal. Results of such detection can be used to make decisions in determination of an adjustment spectrum that can be applied to the power spectrum. The adjusted spectrum can be transformed back into a time-domain signal that substantially removes undesirable noise(s) and/or accounts for known sound components such as the network tone.

Abstract: A speaker is disclosed that can be positioned as either a right or left front surround or a right or left rear surround in any surround sound system. The speaker includes a housing including a first driver, a second driver, a third driver, and a fourth driver. A high pass filter is connected with the first and third drivers and the first and third drivers are wired having opposite polarities from one another such that the first driver is out of phase with the third driver by 180 degrees. A low pass filter is connected with a lattice filter, wherein the low pass filter is configured to shift a phase of an input signal ?90 degrees and the lattice filter is configured to shift the phase by adding +45 degrees thereby forming a ?45 degree phase shift. An output of the lattice filter is connected with the second and fourth drivers.

Abstract: Earphone apparatus (40) comprising: a substantially planar substrate (43) defining at least one electrical connection path (44); an electro-acoustic driver (41) and sensing microphone (42) each mounted on the substrate (43) and connected to the at least one electrical connection path (44); wherein the substrate (43) at least in part defines an acoustic waveguide (47) having a part extending through the substrate (43) for conveying sound from outside of the earphone apparatus (40) to the sensing microphone (42); and the part of the acoustic waveguide (47) extends through the substrate (43) substantially normal to the thickness of the substrate (43).

Abstract: Provided are a method and apparatus for reproducing sound. A sound signal may be generated by receiving sound around the apparatus. An output signal may be generated by performing a process according to functions of the apparatus on a residual signal obtained by canceling a feedback signal from the sound signal.

Abstract: Frequency response correction systems and methods for consumer electronic (CE) devices are disclosed. Applying one or more of disclosed techniques can improve a consumer's listening experience. The frequency response of a CE device can be corrected to remove or attenuate salient, undesirable features of with minimal user interaction. Audio quality can be tuned for optimal or near optimal performance even at maximum or near maximum volume levels substantially without fluctuations, clipping, or any other distortions. In some embodiments, audio response of the CE device can be captured, smoothed, and corrected. Correction parameters can be stored on the CE device.

Abstract: Systems, methods and apparatus for capturing at least one audio signal using a plurality of microphones that generate a plurality of representations of the at least one audio signal. In some embodiments, the plurality of microphones are disposed in a multiple-microphone setting so that the at least one audio signal is captured by at least two of the plurality of microphones. In some embodiments, at least one of the plurality of microphones is a microphone of a mobile device. The plurality of representations of the at least one audio signal may be processed to obtain a processed representation of the at least one audio signal.

Abstract: Frequency response correction systems and methods for consumer electronic (CE) devices are disclosed. Applying one or more of disclosed techniques can improve a consumer's listening experience. The frequency response of a CE device can be corrected to remove or attenuate salient, undesirable features of with minimal user interaction. Audio quality can be tuned for optimal or near optimal performance even at maximum or near maximum volume levels substantially without fluctuations, clipping, or any other distortions. In some embodiments, audio response of the CE device can be captured, smoothed, and corrected. Correction parameters can be stored on the CE device.

Abstract: Technologies are generally described for a method for measuring a quality of an audio signal in a mobile device. In some examples, the mobile device includes a receiving unit configured to receive an audio signal transmitted from another device; an audio quality measuring unit configured to measure a quality of the received audio signal; and a transmission unit configured to transmit the measured quality of the audio signal to the another device.

Abstract: An apparatus comprises a volume-control user interface, a transmitter, an audio amplifier (having a corresponding amplification range), and a control circuit operably coupled to the foregoing components. This control circuit is configured to respond to a volume change made via the volume-control user interface that is within the amplification range of the audio amplifier by making a corresponding change to a volume level for an audio signal being transmitted by the transmitter. The control circuit is also configured to respond to a volume change made via the volume-control user interface that exceeds the amplification range of the audio amplifier by transmitting a volume-increase command.

Abstract: An ambulatory medical device is presented. The ambulatory medical device comprises a function module to provide the intended functionality of the device, a controller module to control the device, a sound generation module with an acoustic transducer to produce an acoustic signal and a signal generator to drive the acoustic transducer with a certain frequency. The signal generation module is arranged within a housing of the device. A tuning module varies the frequency used by the signal generator to drive the acoustic transducer and determines one or more frequencies that correspond to an optimum sound level of the acoustic signal outside of the housing of the device. The optimum sound level of the acoustic signal is a maximum sound level outside of the housing of the device, and/or a maximum perceivable sound level as determined by a user.

Abstract: A multimedia acoustics system having the audio frequency digital interface, which includes at least a USB or IEEE1394 interface, a USB or IEEE1394 outlay sound card, a set of D type or T type audio frequency power amplifier and controlling circuit or MPU chip which match it, a PCB board and at least more than a pair of sound box. In every sound box configuring a full band loudspeaker and/or bandpass heavy undertone loudspeaker whose caliber is no more than 7 inches, the loudspeaker is configured a single diaphragm more magnet gaps and more loops which have resistance load characteristic or approximately resistance load characteristic, thereby form audio frequency digital multimedia acoustics system which has super high sensitivity and high fidelity quality and has 2.0 or 2.1 channel which is supplied by pc USB or IEEE1394 interface and 4.1˜7.1 channel which is supplied by single power supply.