Abstract: A heart rate detection apparatus includes a signal acquisition device for acquiring an intra-aural vibration wave signal and converting the intra-aural vibration wave signal into an intra-aural vibration electrical signal; and an arithmetic processing device for processing the intra-aural vibration electrical signal by computing to generate heart rate information. In the prior art, the heart rate detection apparatus cannot accurately and reliably detect the heart rate information due to the weak optical signal detected by a sensor. The heart rate detection apparatus acquires the intra-aural vibration wave signal to obtain the heart rate information, thus enabling the apparatus to accurately and reliably detect heart rate information.

Abstract: A general-purpose audio connector is provided. The audio connector includes an earphone plug that includes a plurality of first terminals, and an earphone connector that includes a plurality of second terminals. The earphone plug is configured to be inserted into and to connect to the earphone connector. The number of poles of the plurality of first terminals that connect to the plurality of second terminals is variable depending on a movement of the earphone plug.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone is provided proximate the speaker to measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. The anti-noise signal is adaptively generated to minimize the ambient audio sounds at the error microphone. A processing circuit that performs the adaptive noise canceling (ANC) function also filters one or both of the reference and/or error microphone signals, to bias the adaptation of the adaptive filter in one or more frequency regions to alter a degree of the minimization of the ambient audio sounds at the error microphone.

Abstract: A method of analyzing performance of frequency lowering hearing aids. The method includes generating a sequentially of noise signals and transmitting acoustical sounds from a sound output device in response to the sequence of noise signals. A sound input device records the acoustical sounds and saves as a first device data. The sound input device with a frequency lowering hearing aid records the acoustical sounds and save as a second device data. The second device data is compared to the first device data and, in response to the comparison, at least one function of the frequency lowering hearing aid is optionally adjusted.

Abstract: A platform for audio and electronic music applications where the electronics are implemented as modules, and the modules mount in a cabinet with a common power supply and infrastructure. The platform addresses problems in electrical, mechanical, usability, and power distribution areas, and is suitable for guitar effects, synthesizers, studio equipment, and DJ gear.

Abstract: A method for detecting whistling in an audio system includes determining an average frequency of an input signal of the audio system, sampling the input signal in consecutive blocks of at least one sample, wherein the average frequency is determined blockwise, and determining whether feedback related whistling is present in the input signal of the audio system by evaluating a stability of the average frequency, wherein the evaluation of the stability of the average frequency comprises: determining a difference of two values of the determined average frequency for two blocks, and comparing the determined difference to a first threshold value.

Abstract: A low power sound recognition sensor is configured to receive an analog signal that may contain a signature sound. Sparse sound parameter information is extracted from the analog signal. The extracted sparse sound parameter information is processed using a sound signature database stored in the sound recognition sensor to identify sounds or speech contained in the analog signal, wherein the sound signature database comprises a plurality of sound signatures each representing an entire word or multiword phrase.

Abstract: An active noise control arrangement has a signal input (SI), a microphone input (MI), a signal output (SO) and a digital interface (DI). A signal processing block (SP) coupled to the microphone input (MI) by means of an amplifier (MA) has a digitally adjustable gain and comprises combining means (CM) and a filter (TP). The signal processing block (SP) is configured to generate an output signal at the signal output (SO) as a function of an input signal at the signal input (SI) and an amplified microphone signal. A control block (CB) is coupled to the digital interface (DI) and configured to adjust the gain of the amplifier (MA).

Abstract: A device and method for controlling reproduction of an audio signal is provided, wherein the device is operated by means of an energy storage device. The method comprises the steps of deactivating a normal mode and activating an energy saving mode. Power consumption from the energy storage device for reproduction of the audio signal is reduced in the energy saving mode when compared to the normal mode. The method comprises reducing in the energy saving mode, a bass frequency component of a frequency spectrum of the audio signal and outputting the audio signal with reduced bass frequency component. The method further comprises ascertaining a charge state of the energy storage device and controlling the reduction in the bass frequency component based on a decrease in the charge state of the energy storage device.

Abstract: A sound processing apparatus includes a processing unit that is configured to acquire an audio signal, perform a correction processing on the acquired audio signal and output the correction-processed audio signal to a sound emitting unit. The correction processing includes an indirect sound adjusting processing in which a given signal processing is performed on an audio signal so as to adjust an influence of an indirect sound to be heard at a sound receiving point, and a frequency characteristic adjusting processing in which a frequency characteristic of an audio signal is adjusted. In the correction processing, a frequency characteristic for the frequency characteristic adjusting processing is determined based on a frequency characteristic of the indirect sound adjusting processing.

Abstract: An earphone line control device includes a power supply management unit, a filter unit, a switching circuit and a control unit. The switching circuit includes a switch in series with a resistance, a first end of the power supply management unit and an end of the filter unit are both connected to a power supply, a second end of the power supply management unit is connected to a first end of the control unit, the other end of the filter unit is connected to a second end of the control unit, a third end of the power supply management unit is connected to ground, a third end of the control unit is connected to ground, an end of the resistance is connected to the power supply, the other end of the resistance is in series with an end of the switch and connected to a fifth end of the control unit, and the other end of the switch is connected to a forth end of the control unit. When the switch is turned off, the control unit outputs a preset coded pulse signal.

Abstract: A communication apparatus and a sound playing method thereof are provided. The method is adapted for a communication apparatus having a first speaker used as an earpiece and a second speaker, in which the first speaker and the second speaker are disposed at opposite sides of the communication apparatus. In the method, an event with a sound notification is detected. A left channel sound and a right channel sound of the sound notification are mixed to generate a mixed sound in response to detecting the event. The mixed sound is played by the first speaker with a fade-in effect.

Abstract: One or more embodiments describe controlling audio signals at a user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in a first or a second mode. In dependence upon determining that the user device is operating in the first mode, the secondary audio signals are selectively suppressed from being output from the user device during the communication session.

Abstract: A glass loudspeaker emitting sound and acoustically-driven light includes a glass substrate, a plurality of exciters, a light bar, a plurality of coupling collimators, a diffractive grating, and a control unit. The exciters are arranged at intervals on the glass substrate. The light bar is arranged toward the glass substrate and includes a plurality of light sources. The coupling collimators are arranged on the light-emitting path of the light sources. The diffractive grating is arranged on the glass substrate and toward the light bar. The control unit is electrically connected with the light bar and the exciters, and includes an acoustic signal receiving unit, an audio signal amplifier, an audio signal analyzer, and a current distributor.

Abstract: A portable audio device, which includes active noise cancellation circuitry, a hearing aid compliant magnetic radiator, and a speaker/earpiece, is surrounded by ambient acoustic noise. The active noise cancellation circuitry provides an anti-noise signal at an input of the speaker to control/reduce the ambient acoustic noise outside of the device. In addition, the active noise cancellation circuitry provides an inverse anti-noise signal to an input of the magnetic radiator. The magnetic fields produced by the speaker driven by the anti-noise signal and the magnetic radiator driven by the inverse anti-noise signal cancel each other out through phase cancellation such that a hearing aid using a telecoil coupled to the audio device does not produce significant audio waves based on either of these signals. Other embodiments are also described.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: Responsive to the absence of a reference clock input signal to a clock conditioning circuit for generating a desired clock signal for synchronizing components of an audio signal path, a controller may cause the signal path to receive at the clock input a substitute clock signal in the absence of the reference clock input signal and may modify one or more parameters of the signal path in order to perform one or more of the following: (i) reduce the presence of audio artifacts in the output signal caused by the absence of the reference clock input signal; (ii) power down at least one component of the signal path to reduce power consumed by the signal path; (iii) continue to operate the signal path with the substitute clock signal; and (iv) transition the signal path to a mute condition or a zero volume condition.

Abstract: Systems and processes for compensating for changes in a theatre sound system positioned in a theatre are described. A subsequent response of a loudspeaker to a test signal is captured and compared to a previously obtained signature response of the loudspeaker to the test signal. An audio signal can be processed based on the comparison to compensate for changes to loudspeaker performance, or otherwise.

Abstract: An encoding method and apparatus and a decoding method and apparatus are provided. The decoding method includes skipping extension information included in an input bitstream, extracting a three-dimensional (3D) down-mix signal and spatial information from the input bitstream, removing 3D effects from the 3D down-mix signal by performing a 3D rendering operation on the 3D down-mix signal, and generating a multi-channel signal using a down-mix signal obtained by the removal and the spatial information. Accordingly, it is possible to efficiently encode multi-channel signals with 3D effects and to adaptively restore and reproduce audio signals with optimum sound quality according to the characteristics of an audio reproduction environment.

Abstract: Systems and methods are provided to determine a subset of D microphones in a set of N microphones on a perimeter of a space to monitor a target location. The space is divided into L interference locations. An equation is solved to determine microphone weights for the N microphones by minimizing the maximum gain for signals related to the target location and interference locations, further optimized over an l1 penalty by applying a Lagrange multiplier to an l1 norm of the microphone weights in a manner that determines a set of D non-zero microphones weights and a set of (N-D) microphone weights that are zero or close to zero. Microphone weights are determined for at least 2 different frequencies.